2012-07-05 Asterisk Development Team * Asterisk 10.5.2-digiumphones Released. * AST-2012-010 * AST-2012-011 2012-06-14 Asterisk Development Team * Asterisk 10.5.1-digiumphones Released. * AST-2012-009 2012-06-04 Asterisk Development Team * Asterisk 10.5.0-digiumphones Released. 2012-05-30 Asterisk Development Team * Asterisk 10.5.0-digiumphones-rc2 Released. * Resolve crash in subscribing for MWI notifications. ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable shoudl definitely not be used after that. To solve this in the two cases that affect subscribing for MWI notifications, we instead save the ref locally, and unref them in the error conditions. (closes issue ASTERISK-19827) Reported by: B. R. Review: https://reviewboard.asterisk.org/r/1940/ * Fix crash in ConfBridge when user announcement is played for more than 2 users A patch introduced in r354938 made it so that ConfBridge would not attempt to play sound files if those files did not exist. Unfortunately, ConfBridge uses the same underlying fucntion, play_sound_helper, to playback both the sound files and numbers to callers. When a number is being played back, the name of the sound file is expected to be NULL. This NULL value was passed into a function that tested for the existance of a sound file and is not tolerant to NULL file names, causing a crash. This patch fixes the behavior, such that if a sound file does not exist we do not attempt to play it, but we only attempt that check if the sound file was specified in the first place. If a sound file was not specified, we use the 'play number' logic in the helper function. (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested by: Florian Gilcher patches: ASTERISK-19899.diff uploaded by mjordan (license 6283) * AST-2012-007 * AST-2012-008 2012-05-03 Asterisk Development Team * Asterisk 10.5.0-digiumphones-rc1 Released. 2012-05-03 20:06 +0000 [r365264] Jason Parker * main/jitterbuf.c, configs/queues.conf.sample, configs/usbradio.conf.sample (removed), res/res_calendar_caldav.c, apps/rpt_flow.pdf (removed), apps/app_queue.c, main/cel.c, res/res_config_sqlite.c, res/res_calendar_ews.c, main/config.c, formats/format_siren7.c, channels/chan_dahdi.c, formats/format_vox.c, funcs/func_volume.c, configure, formats/format_h263.c, main/event.c, apps/app_chanspy.c, formats/format_g719.c, channels/chan_sip.c, funcs/func_env.c, channels/chan_agent.c, funcs/func_strings.c, channels/console_video.c, Makefile.rules, main/astfd.c, formats/format_wav_gsm.c, bridges/bridge_multiplexed.c, channels/chan_iax2.c, funcs/func_global.c, apps/confbridge/conf_config_parser.c, res/res_config_curl.c, build_tools/cflags.xml, main/cdr.c, funcs/func_curl.c, main/manager.c, main/tdd.c, channels/console_gui.c, formats/format_pcm.c, main/app.c, main/stdtime/localtime.c, utils/extconf.c, makeopts.in, main/message.c, formats/format_gsm.c, res/res_clioriginate.c, include/asterisk/time.h, res/res_rtp_asterisk.c, res/res_config_pgsql.c, apps/app_meetme.c, /, formats/format_wav.c, configure.ac, res/res_musiconhold.c, channels/chan_gtalk.c, tests/test_linkedlists.c, apps/app_ices.c, channels/sig_pri.c, res/res_srtp.c, formats/format_ilbc.c, channels/sig_pri.h, Makefile, apps/app_forkcdr.c, res/res_config_odbc.c, bridges/bridge_builtin_features.c, codecs/gsm/src/k6opt.s, build_tools/menuselect-deps.in, funcs/func_channel.c, apps/app_directed_pickup.c, main/features.c, res/res_agi.c, main/http.c, main/logger.c, apps/app_confbridge.c, apps/app_sms.c, main/audiohook.c, formats/format_h264.c, apps/app_voicemail.c, codecs/lpc10/Makefile, apps/app_dial.c, formats/format_sln.c, codecs/gsm/Makefile, funcs/func_sysinfo.c, formats/format_ogg_vorbis.c, CHANGES, main/astobj2.c, main/format_pref.c, apps/app_speech_utils.c, tests/test_security_events.c, main/tcptls.c, addons/ooh323cDriver.c, formats/format_g723.c, apps/app_externalivr.c, tests/test_config.c, tests/test_poll.c, addons/chan_mobile.c, formats/format_siren14.c, funcs/func_devstate.c, main/asterisk.c, main/xmldoc.c, channels/chan_mgcp.c, formats/format_g729.c, channels/chan_unistim.c, configs/chan_dahdi.conf.sample, main/pbx.c, res/res_calendar_icalendar.c, channels/chan_local.c, funcs/func_version.c, configs/rpt.conf.sample (removed): Multiple revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 ........ r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr 2012) | 10 lines Make 'help devstate change' display properly (get rid of excess comma) (closes issue ASTERISK-19444) Reported by: Makoto Dei Patches: devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr 2012) | 12 lines Fix some stuff involving calls to memcpy and memset The important parts of the patch were already applied through other updates. (closes issue ASTERISK-19445) Reported by: Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) | 10 lines Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined There were a few instances of restarting music on hold in meetme that would cause Asterisk to revert to the default class of music on hold for no adequate reason. Review: https://reviewboard.asterisk.org/r/1844/ ........ Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) | 11 lines Remove unnecessary error message in app_dial.c The error message for failure to stop autoservice after a gosub or macro call during a dial was removed for macro while Asterisk 1.4 was still being actively developed. The corresponding gosub error message was never removed. (closes issue ASTERISK-19551) ........ Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012) | 11 lines Fix a typo in the warning messages for an ignored media stream Added a '\n' to the warning messages when we ignore a media stream due to the port number being '0'. (closes issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........ Merged revisions 361332 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012) | 5 lines Remove a few more files related to chan_usbradio and app_rpt. ........ Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr 2012) | 14 lines Multiple revisions 361403,361412 ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ Merged revisions 361403,361412 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) | 5 lines Add missing newlines to CLI logging ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012) | 8 lines Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key header did not already exist. If it already existed it would delete it. * Made msg_set_var_full() exit early if the named variable did not already exist and the value to set is empty. ........ r361560 | mjordan | 2012-04-06 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines Fix memory leak when using MeetMeAdmin 'e' option with user specified A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command (eject last user that joined) is used in conjunction with a specified user. Regardless of the command being executed, if a user is specified for the command, MeetMeAdmin will look up that user. Because the 'e' option kicks the last user that joined, as opposed to the one specified, the reference to the user specified by the command would be leaked when the user variable was assigned to the last user that joined. ........ Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06 Apr 2012) | 12 lines Fix memory leak in res_calendar_ews when event email address node is empty If the XML calendar data returned by a Microsoft Exchange Web Service specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address is provided, a condition existed where an ast_calendar_attendee struct would be allocated but not appended to the list of attendees. Because of that, the memory associated with the attendee would never be freed. This patch frees the memory if no e-mail address is provided. ........ Merged revisions 361606 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012) | 15 lines Change SHARED function to use a safe traversal when modifying a variable When the SHARED function modifies a variable, it removes it from its list of variables and reinserts the new value at the head of the list of variables. Doing this inside a standard list traversal can be dangerous, as the standard list traversal does not account for the list being changed. While the code in question should not cause a use after free violation due to its breaking out of the loop after freeing the variable, it could lead to a maintenance issue if the loop was modified. This also fixes a violation reported by a static analysis tool, which also makes this code easier to maintain in the future. ........ Merged revisions 361657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012) | 17 lines Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. This patch makes it so that we only free the caller ID structure if a DAHDI channel is successfully created, and we bump the gains back up if we fail to make a DAHDI channel. ........ Merged revisions 361705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012) | 12 lines Allow func_curl to exit gracefully if list allocation fails during write If the global_curl_info data structure could not be allocated, the datastore associated with the operation would be free'd, but the function would not return. This would later dereference the datastore, almost certainly causing Asterisk to crash. With this patch, if the data structure is not allocated the method will return an error code, and not attempt any further operation. ........ Merged revisions 361753 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012) | 10 lines Fix crash caused by unloading or reloading of res_http_post When unlinking itself from the registered HTTP URIs, res_http_post could inadvertently free all URIs registered with the HTTP server. This patch modifies the unregister method to only free the URI that is actually being unregistered, as opposed to all of them. ........ Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012) | 19 lines Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. * Rework the -r361705 patch to better manage the cs and mtd allocated resources. * Fixed use of mwimonitoractive flag to be correct if the mwi_thread() fails to start. ........ Merged revisions 361854 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) | 10 lines Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8 Prior to this patch, in order to restore that behavior, a function would have to be used on the QueueMember to make the ringinuse option do anything, which is pretty unreasonable. (closes issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1860/ ........ r361956 | kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines Simplify build system architecture optimization This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) ........ Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) | 12 lines Make trunkfreq take effect when set Previously, setting trunkfreq had no effect on initial load or on reload and only ever used the default value. This causes trunkfreq to be used appropriately on initial load and reload. (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........ Merged revisions 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr 2012) | 10 lines Send relative path named recordings to the meetme directory instead of sounds Prior to this patch, no effort was made to parse the path name to determine a proper destination for recordings of MeetMe's r option. This fixes that. Review: https://reviewboard.asterisk.org/r/1846/ ........ Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) | 15 lines Make ForkCDR e option not set end time of the newly forked CDR log Prior to this patch, ForkCDR's e option would immediately set the end time of the forked CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time being roughly the same as it's beginning time (which is in turn roughly the same as the original's end time). (closes issue ASTERISK-19164) Reported by: Steve Davies Patches: cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012) ........ Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012) | 19 lines Check for IO stream failures in various format's truncate/seek operations For the formats that support seek and/or truncate operations, many of the C library calls used to determine or set the current position indicator in the file stream were not being checked. In some situations, if an error occurred, a negative value would be returned from the library call. This could then be interpreted inappropriately as positional data. This patch checks the return values from these library calls before using them in subsequent operations. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012) | 18 lines Fix handling of negative return code when storing voicemails in ODBC storage When storing a voicemail message using an ODBC connection to a database, the voicemail message is first stored on disk. The sound file associated with the message is read into memory before being transmitted to the database. When this occurs, a failure in the C library's lseek function would cause a negative value to be passed to the mmap as the size of the memory map to create. This would almost certainly cause the creation of the memory map to fail, resulting in the message being lost. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863 ........ Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012) | 25 lines Fix negative return handling in channel drivers In chan_agent, while handling a channel indicate, the agent channel driver must obtain a lock on both the agent channel, as well as the channel the agent channel is using. To do so, it attempts to lock the other channel first, then unlock the agent channel which is locked prior to entry into the indicate handler. If this unlock fails with a negative return value, which can occur if the object passed to agent_indicate is an invalid ao2 object or is NULL, the return value is passed directly to strerror, which can only accept positive integer values. In chan_dahdi, the return value of dahdi_get_index is used to directly index into the sub-channel array. If dahd_get_index returns a negative value, it would use that value to index into the array, which could cause an invalid memory access. If dahdi_get_index returns a negative number, we now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012) | 23 lines Turn off warning message when bind address is set to any. When a bind address is set to an ANY address (udpbindport=::), a warning message is displayed stating that "Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr' settings." But if one is running dual stack, we shouldn't be told to turn those settings off. This patch checks if the bind address is an ANY address or not. The warning message will now only be displayed if the bind address is NOT an ANY address and IPv6 is being used. Also, updated the copyright year. (closes issue ASTERISK-19456) Reported by: Michael L. Young Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012) | 15 lines Fix error that caused seek format operations to set max file size to '1' or '0' A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan ........ Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012) | 17 lines Fix places where a negative return from ftello could be used as invalid input In a variety of locations in both reading and writing a file, the result from the C library function ftello is used as input to other functions. For the parameters and functions in question, a negative value is invalid input. This patch checks the return value from the ftello function to determine if we were able to determine the current position in the file stream and, if not, fail gracefully. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362355 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) | 12 lines Make use of va_args more appropriate to form in various res_config modules plus utils. A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ ........ Merged revisions 362354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012) | 24 lines Fix places in main where a negative return value could impact execution This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012) | 29 lines Fix places in resources where a negative return value could impact execution This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012) | 13 lines Handle case where an unknown format is used to get the preferred codec size In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18 Apr 2012) | 19 lines Add ability to ignore layer 1 alarms for BRI PTMP lines. Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18 Apr 2012) | 50 lines Fix a variety of potential buffer overflows * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012) | 14 lines Handle multiple commands per connection via netconsole Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr 2012) | 12 lines Prevent a crash in ExternalIVR when the 'S' command is sent first. If the first command sent from an ExternalIVR client is an 'S' command, we were blindly removing the first element from the play list and deferencing it, even if it was NULL. This corrects that and also locks appropriately in one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski ........ Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012) | 5 lines Update membermacro and membergosub documentation in queues.conf.sample. ........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012) | 9 lines Add leading and trailing backslashes A couple of unit tests did not have have leading or trailing backslashes when setting their test category resulting in a warning message being displayed. Added the backslash where needed. ........ Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012) | 5 lines Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012) | 13 lines Document Speech* apps hangup on failure and suggest TryExec The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) ........ Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012) | 11 lines OpenBSD doesn't have rawmemchr, use strchr (closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012) | 11 lines Add missing payload type to events API The Security Events Framework API was changed while adding the generation of security events in chan_sip. A payload type and name was missed from being added to struct ie_maps. (closes issue ASTERISK-19759) Reported by: Michael L. Young Patches: issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026) ........ r362998 | rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines Update app_dial M and U option GOTO return value documentation. ........ Merged revisions 362997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012) | 8 lines On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012) | 5 lines Hangup affected channel in error paths of bridge_call_thread(). ........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012) | 27 lines Fix recalled party B feature flags for a failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012) | 19 lines Clear ISDN channel resetting state if the peer continues to use it. Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012) | 18 lines Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012) | 5 lines Update Pickup application documentation. ........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012) | 5 lines Update Pickup application documentation. (Even better) ........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr 2012) | 14 lines chan_sip: [general] maxforwards, not checked for a value greater than 255 The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) | 15 lines Fix reference leaks involving SIP Replaces transfers The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Karjewski ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012) | 8 lines Add more constness to the end_buf pointer in the netconsole issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012) | 24 lines Fix DTMF atxfer running h exten after the wrong bridge ends. When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario ........ Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012) | 5 lines Update Pickup application documentation. (With feeling this time.) ........ Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012) | 3 lines fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time. ........ r364204 | mjordan | 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines Allow for reloading SRTP crypto keys within the same SIP dialog As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27 Apr 2012) | 14 lines Allow SIP pvts involved in Replaces transfers to fall out of reference sooner Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Krajewski ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012) | 43 lines Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr 2012) | 10 lines Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails. (closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) ........ Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines Add missing test_config.c ........ r364536 | elguero | 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines Fix configuring custom sound_leader_has_left in confbridge.conf The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ ........ r364579 | mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines Fix error that caused truncate operations to fail Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) | 15 lines Merged revisions 364635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ r364651 | may | 2012-04-30 11:48:57 -0500 (Mon, 30 Apr 2012) | 10 lines Fix use freed pointer in return value from call thread (issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) | 13 lines Fix bad check in voicemail functions for ast_inboxcount2_func Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) | 12 lines Play conf-placeintoconf message to the correct channel Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012) | 7 lines * Fix error path resouce leak in local_request(). * Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May 2012) | 16 lines Fix Coverity-reported ARRAY_VS_SINGLETON error. As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012) | 10 lines Fixed __ao2_ref() validating user_data twice. (closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012) | 11 lines Only log a failure to get read/write samples from factories if it didn't happen In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ r365014 | elguero | 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ ........ r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02 May 2012) | 33 lines Multiple revisions 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-02 Asterisk Development Team * Asterisk 10.4.0 Released. 2012-05-01 Asterisk Development Team * Asterisk 10.4.0-rc3 Released. * channels/chan_sip.c: Revert revision 360862 Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) 2012-04-24 Asterisk Development Team * Asterisk 10.4.0-rc2 Released. * AST-2012-004 * AST-2012-005 * AST-2012-006 2012-04-04 Asterisk Development Team * Asterisk 10.4.0-rc1 Released. 2012-04-04 16:38 +0000 [r361091-361143] Jonathan Rose * main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /, channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c, channels/chan_iax2.c, res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU old-style field designator extensions to fix clang warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) ........ Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8 Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue ASTERISK-19540) * /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ ........ Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-03 20:08 +0000 [r360993-361041] Kinsey Moore * /, apps/app_transfer.c: Fix the display of documentation for Transfer This came up while fixing documentation generation for many other cases where the argument separator was not being displayed properly. Now that it is displayed properly, it shows up in the wrong place for Transfer since the '/' is only required if Tech is present. (related to issue ASTERISK-18168) ........ Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) ........ Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-30 21:29 +0000 [r360934] Richard Mudgett * /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan ........ Merged revisions 360933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-29 23:33 +0000 [r360863-360885] Mark Michelson * /, main/features.c: Fix potential race condition during call pickup. Prior to this patch, a connected line update was queued during call pickup and then an answer frame was queued. The original caller would presumably then have his connected line updated and then the call would be answered. In actuality, the answer frame was not how the call ended up being answered. Rather, an odd section in app_dial that checks if the called channel's state is up. The result is that the order of the connected line update and the answer were variable. In most cases, this wasn't actually a bad thing. However, if the 'I' option was passed to dial, the connected line update would be inhibited. The fix is to queued the connected line after the answer frame is queued. This way the race in app_dial is between two conditions resulting in an answer. This way the connected line update occurs after the answer every time. (closes issue ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by Mark Michelson (license 5049) ........ Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests. This change makes use of connected party information in addition to caller ID in order to populate local and remote XML elements in the dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej Krajewski Tested by: Maciej Krajewski Patches: local_remote_hint2.diff uploaded by Mark Michelson (license 5049) ........ Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-28 19:20 +0000 [r360717] Terry Wilson * channels/chan_jingle.c, addons/chan_ooh323.c, /, cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, channels/chan_gtalk.c: Destroy configs when they are no longer used https://reviewboard.asterisk.org/r/1834/ ........ Merged revisions 360712 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-27 18:23 +0000 [r360672] Mark Michelson * /, channels/chan_sip.c: Make a debug message regarding subscription changes more accurate. I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. ........ Merged revisions 360625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-27 14:35 +0000 [r360489-360575] Jonathan Rose * /, configure: Updates config with bootstrap where I changed configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon Clark ........ Merged revisions 360574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configure.ac: Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-26 18:41 +0000 [r360472-360476] Paul Belanger * /, CHANGES: Update CHANGES for r360471 ........ Merged revisions 360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/dnsmgr.c, /: Increase verbosity level for ast_verb messages While this does not fix the issue of the CLI being flooded by 'doing dnsmgr_lookup' messages, increasing the verbosity level above 5 should help minimize it. ........ Merged revisions 360471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-24 23:47 +0000 [r360358-360414] Russell Bryant * funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error handling code path. ........ Merged revisions 360413 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_page.c: app_page: Fix a memory leak on every Page(). dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. ........ Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_jack.c, /: app_jack: fix datastore memory leak in error handling path. ........ Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/ast_expr2.c, /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h: Multiple revisions 360356-360357 ........ r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines expression parser: Fix (theoretical) memory leak. Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. ........ r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines Rebuild parsers. This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. ........ Merged revisions 360356-360357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-24 00:37 +0000 [r360263-360310] Richard Mudgett * main/channel.c, /, channels/sig_pri.c: Make number not available presentation also set screening to network provided. Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Add missing initialization of update_redirecting in chan_sip.c ........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-21 14:52 +0000 [r360139] Jonathan Rose * contrib/scripts/install_prereq, /: Update install_prereq script to include missing GSM library for debian amd move SQLite3. (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 360138 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-21 14:21 +0000 [r360098] Tzafrir Cohen * /, configure, configure.ac: Also detect gmime 2.6 Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) ........ Merged revisions 360087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-21 13:28 +0000 [r360088] Matthew Jordan * /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ Merged revisions 360086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-20 20:37 +0000 [r360034] Kinsey Moore * /, apps/app_echo.c: Prevent Echo() from relaying control, null, and modem frames Echo()'s description states that it echoes audio, video, and DTMF except for # while it actually echoes any frame that it receives other than DTMF #. This was causing frame storms in the test suite in some circumstances where Echo() was attached to both ends of a pair of local channels and control frames were being periodically generated. Echo()'s behavior and description have been modifed so that it only echoes media and non-# DTMF frames. ........ Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-20 18:11 +0000 [r359982] Sean Bright * channels/chan_iax2.c: chan_iax2: Emit Port alongside Post in PeerStatus AMI Event. The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. So include Port along with Post when emitting the event. We'll remove Post in trunk. 2012-03-20 17:25 +0000 [r359980] Richard Mudgett * main/manager.c, /, include/asterisk/manager.h: Allow AMI action callback to be reentrant. Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ ........ Merged revisions 359979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-16 20:20 +0000 [r359898] Jonathan Rose * /, apps/app_chanspy.c: Prevent chanspy from binding to zombie channels This patch addresses a bug with chanspy on local channels which roughly 50% of the time would create a situation where chanspy can latch onto a zombie channel, keeping the zombie alive forever and causing the channel doing the spying to never be able to hang up. (closes issue ASTERISK-19493) Reported by: lvl Review: https://reviewboard.asterisk.org/r/1819/ ........ Merged revisions 359892 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-16 08:24 +0000 [r359810] Alec L Davis * /, channels/sip/include/sip.h: Missed lastinvite CSeq int to uint32_t change from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-15 19:06 +0000 [r359694-359707] Matthew Jordan * /, main/utils.c: Fix remotely exploitable stack overflow in HTTP manager There exists a remotely exploitable stack buffer overflow in HTTP digest authentication handling in Asterisk. The particular method in question is only utilized by HTTP AMI. When parsing the digest information, the length of the string is not checked when it is copied into temporary buffers allocated on the stack. This patch fixes this behavior by parsing out pre-defined key/value pairs and avoiding unnecessary copies to the stack. (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested by: Matt Jordan ........ Merged revisions 359706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun in Milliwatt Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it ........ Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-15 18:22 +0000 [r359620] Richard Mudgett * apps/app_dial.c, /, apps/app_queue.c: Add missing connected line macro calls to initial dial for Dial and Queue apps. The connected line interception macros do not get executed when the outgoing channel is initially created and that channel's caller-id is implicitly imported into the incoming channel's connected line data. If you are using the interception macros, you would expect that they get run for every change to a channel's connected line information outside of normal dialplan execution. Review: https://reviewboard.asterisk.org/r/1817/ ........ Merged revisions 359609 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-15 00:53 +0000 [r359454-359559] Russell Bryant * /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in try_transfer() so that the code isn't (potentially) trying to read from it while uninitialized. ........ Merged revisions 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of uninitialized variable. Avoid potential use of idroster in gtalk_alloc() before it has been initialized. ........ Merged revisions 359508 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_chanisavail.c: app_chanisavail: Fix use of uninitialized variable. Ensure that status is set before it is used by resetting it during each loop iteration. This could have resulted in incorrect results from this app. ........ Merged revisions 359486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is initialized. Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). ........ Merged revisions 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * include/asterisk/app.h, /: app.h: Always initialize AST_DECLARE_APP_ARGS(). This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always fully initialized. I'm not sure if this fixes any real bugs, but it silences a bunch of warnings from coverity, and is generally a good thing to do anyway. ........ Merged revisions 359452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 22:28 +0000 [r359453] Richard Mudgett * main/channel.c, /, channels/chan_agent.c, include/asterisk/channel.h: Fix deadlock potential with some ast_indicate/ast_indicate_data calls. Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 17:42 +0000 [r359358] Matthew Jordan * /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to missed resynchronizations When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) ........ Merged revisions 359356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 17:24 +0000 [r359355] Richard Mudgett * apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and forked calls generating warnings for voice frames. When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ ........ Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 10:54 +0000 [r359051-359260] Russell Bryant * include/asterisk/logger.h, /, main/logger.c: Fix bogus reads/writes of console log levels in asterisk.c This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. ........ Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid reads/writes due to incorrect sizeof(). These few places in the code used sizeof() on h_addr in struct hostent. This is sizeof(char *). The correct way to get the size of this address is to use h_length. This error would result in reads/writes of 8 bytes instead of 4 on 64-bit machines. ........ Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/sched.c, /: Fix inaccurate sizeof() in sched.c. This code just needed sizeof(int), not sizeof(int *). ........ Merged revisions 359157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, utils/astman.c: Fix incorrect sizeof() in astman. ........ Merged revisions 359116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_crypto.c: Fix incorrect usage of sizeof() in res_crypto. In this case, just remove the memset(). There was a redundant memset that is done correctly just 2 lines later. ........ Merged revisions 359110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi. ........ Merged revisions 359088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Fix incorrect sizeof() usage in features.c. This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. ........ Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final(). ........ Merged revisions 359059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/pbx.c, /: Don't use a buffer after it goes out of scope. 's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. ........ Merged revisions 359056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/res_ais.c, /, res/ais/clm.c, res/ais/evt.c, res/ais/ais.h: Dump cache of published events when a node joins the cluster. Also use a more reliable method for stopping the poll() thread. ........ Merged revisions 359053 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_usbradio.c (removed), /, channels/xpmr (removed), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-13 20:36 +0000 [r358944-358989] Terry Wilson * /, main/features.c: Fix setting CDR variables in the hangup extension A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * include/asterisk/devicestate.h, /, channels/chan_sip.c, tests/test_devicestate.c, main/devicestate.c: Make hints for invalid SIP devices return Unavail, not idle This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-13 16:58 +0000 [r358811-358860] Tilghman Lesher * /, UPGRADE.txt, CHANGES: Requested changes documenting the fixed AEL functionality. ........ Merged revisions 358859 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c, utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4. This change restores functionality that was present in 1.4, when AEL macros were implemented with the Macro dialplan application. Macros are fraught with functionality issues, because they consume a large portion of the underlying application stack. This limits the ability of AEL users to call many layers of subroutines, an issue which Gosub does not have (originally tested to 100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were implemented with Gosub. However, there were some implicit behaviors of Macro, which were not replicated at the same time as with the transition to Gosub, one of which is documented in the related issue. In particular, the "h" extension is designed to execute not in the Macro context, but in the topmost calling context. Due to legacy issues with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks in all calling contexts, bubbling up from the deepest level until it finds an "h" extension. Since AEL hides the complexity of the underlying dialplan logic from the AEL programmer, it's reasonable to assume that this behavior should not change in the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break working AEL configurations in the transition to Asterisk 1.8 LTS. This fix is the result, which implements a search for the "h" extension in all calling Gosub contexts. Fixes ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher (with slight modifications for 1.8) Tested by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1776/ ........ Merged revisions 358810 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-08 16:50 +0000 [r358644] Jonathan Rose * /, channels/chan_sip.c: Make transfer not ignore port information with SIP. Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from the host string and ignored. This simply keeps chan_sip from cutting off the port number during these kinds of transfers. (closes issue ASTERISK-19321) Reported by: Federico Alves Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header ........ Merged revisions 358643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-07 18:28 +0000 [r358531] Richard Mudgett * /, channels/sig_ss7.c: Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue ASTERISK-19372) ........ Merged revisions 358530 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-07 16:13 +0000 [r358485] Sean Bright * /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the number of samples set properly. If the wctc4xxp returns more than a single packet, we need to update the number of samples in the returned frame accordingly. Acked-by: Shaun Ruffell ........ Merged revisions 358484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-07 15:17 +0000 [r358436-358441] Terry Wilson * /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for ODBC WCHAR fields Without detecting these types, cel_odbc blows up when the character set for the table is utf8. This also wraps cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the code. ........ Merged revisions 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-06 17:46 +0000 [r358261-358378] Richard Mudgett * channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing calls on FXS ports. * Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for compiling with -Wshadow and finding this bug. ........ Merged revisions 358377 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_ss7.c: Drop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should clear a failed call as soon as possible. * Made SS7 hangup a call immediately if it has not connected yet for INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate inband tone. (closes issue ASTERISK-19372) Reported by: Igor Nikolaev ........ Merged revisions 358278 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: Setup DSP when SS7 call is connected or early media is available. Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 18:58 +0000 [r358215] Jonathan Rose * main/manager.c, /: Eliminate double close of file descriptor in manager.c The process_output function in manager.c attempted to call fclose and close immediately afterwards. Since fclose implies close, this resulted in a potential double free on file descriptors. This patch changes that behavior and also adds error checking to fclose and close depending on which was deemed necessary. Also error messages. Thanks to Rosen Iliev for pointing out the location of the problem. (closes issue ASTERISK-18453) Reported By: Jaco Kroon Review: https://reviewboard.asterisk.org/r/1793/ ........ Merged revisions 358214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 16:42 +0000 [r358163] Joshua Colp * /, channels/chan_sip.c: Defer sending the connected line reinvite if a reinvite is already in progress. (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) ........ Merged revisions 358162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 15:59 +0000 [r358116] Kinsey Moore * /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Patch-by: Jeremiah Gowdy ........ Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 23:28 +0000 [r357987-358033] Terry Wilson * channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix unused-but-set-variable warnings All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. ........ Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Correct some set-but-unused variable warnings in the mISDN library. (from kpfleming's commit to trunk r356292) ........ Merged revisions 358011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode x=++x and x=x=1? Really? ........ Merged revisions 357986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 21:03 +0000 [r357941] Kinsey Moore * /, main/ccss.c, tests/test_event.c, main/event.c, include/asterisk/strings.h: Fix case-sensitivity for device-specific event subscriptions and CCSS This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 18:37 +0000 [r357895] Richard Mudgett * main/channel.c, /, channels/sig_pri.c: Remove ISDN hold restriction for non-bridged calls. The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 15:59 +0000 [r357812] Sean Bright * /, channels/chan_iax2.c: The default value for mohinterpret is the empty string, so when resetting to default values don't explicitly set the value to "default." ........ Merged revisions 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 15:50 +0000 [r357810] Richard Mudgett * /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix next_channel() channel reference leak in ChanSpy. (closes issue ASTERISK-19461) Reported by: Irontec Patches: app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec (issue ASTERISK-17515) ........ Merged revisions 357809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 01:05 +0000 [r357762] Mark Michelson * main/channel.c, /: Fix race condition that can cause important control frames (such as a hangup) to be missed. This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 ........ Merged revisions 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-01 14:18 +0000 [r357667] Kinsey Moore * /, main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) ........ Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-29 20:39 +0000 [r357576-357620] Walter Doekes * include/asterisk/stringfields.h, main/utils.c: Update stringfield documentation for removed second va_list in favor of va_copy. In r320946, the second va_list that was passed to ast_string_field_build_va and friends, was removed. This patch updates the documentation to reflect that. * apps/app_dial.c, /: Fix copying of CDR(accountcode) to local channels. In r203638, during the addition of the Channel Event Logging, in mid-2009, this got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the CDR(accountcode) from the calling channel is available to dialed channels again as well as showing up properly in the CDR's. (closes issue ASTERISK-19384) Patches: accountcode.patch (License #6033) by jamicque Review: https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard Mudgett ........ Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 22:29 +0000 [r357458-357497] Jonathan Rose * /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, cdr/cdr_adaptive_odbc.c: Add additional character type types to supported data types for cdr_adaptive_odbc The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this patch adds those along with some other character types to the list of types cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really UTF8 characters as much as it was a failure to respond to the exact type that was declared/in use on that database. (closes issue ASTERISK-19334) Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236) ........ Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 21:21 +0000 [r357421] Tilghman Lesher * /, apps/app_stack.c: Correctly reset the dialplan priority. When the stack frame is allocated, we save the address to which we should return, when the Gosub returns. However, if we just want to restore the priority, then we need to subtract 1 before setting it. Otherwise, when a Gosub goes to a nonexistent address, it will skip a priority in the dialplan. This is because when we return from an application, the PBX increments the priority for us. ........ Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 20:58 +0000 [r357408] Richard Mudgett * /, channels/sig_pri.c: Use more reasonable cause code when rejecting incoming call waiting calls. (closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter ........ Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 20:42 +0000 [r357357-357405] Jonathan Rose * UPGRADE.txt: revision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352) reported by: jamicque * /, UPGRADE.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12 (issue ASTERISK-19352) reported by: jamicque ........ Merged revisions 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating changes to transport option (issue ASTERISK-19352) Reported by: jamicque ........ Merged revisions 357356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 19:35 +0000 [r357353] Richard Mudgett * /, apps/app_page.c: Remove dupliate 'i' option table entry in app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei Patches: app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei ........ Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 18:51 +0000 [r357318] Mark Michelson * channels/sip/security_events.c: Add a security event for the case where fake authentication challenge is sent. 2012-02-28 18:11 +0000 [r357271] Jonathan Rose * /, channels/chan_sip.c: Changes transport option in sip.conf so that using multiple instances doesn't stack. Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header ........ Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 14:46 +0000 [r357213] Kevin P. Fleming * /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The build system has some special magic to ensure that if Asterisk is built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled with the optimizer enabled (even though the result will be thrown away), because the compiler is able to find a great deal of coding errors and bugs as a result of running its optimizers. Unfortunately at some point this mode got broken, and the 'throwaway' compile of the code was no longer done with the optimizer enabled. This patch corrects that problem. ........ Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-29 Asterisk Development Team * Asterisk 10.3.0 Released. 2012-03-26 Asterisk Development Team * Asterisk 10.3.0-rc3 Released. * AST-2012-003 * AST-2012-002 * /main/manager.c, include/asterisk/manager.h: Fix AMI deadlock regression by allowing AMI action callback to be reentrant Fix AMI module reload deadlock from ASTERISK-18479 when it tired to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13874 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The path stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ 2012-03-06 Asterisk Development Team * Asterisk 10.3.0-rc2 Released. * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0. In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. 2012-01-30 22:16 +0000 [r353369-353321] Alec L Davis * channels/sip/include/dialog.h, /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ * /, channels/chan_sip.c: Merged revisions 353368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 ........ 2012-01-30 23:28 +0000 [r353397] Terry Wilson * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h: Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ ........ Merged revisions 353371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-31 17:21 +0000 [r353463] Richard Mudgett * main/manager.c, /, include/asterisk/channel.h: Fix memory leak in error paths for action_originate(). * Fix memory leak of vars in error paths for action_originate(). * Moved struct fast_originate_helper tech and data members to stringfields. * Simplified ActionID header handling for fast_originate(). * Added doxygen note to ast_request() and ast_call() and the associated channel callbacks that the data/addr parameters should be treated as const char *. Review: https://reviewboard.asterisk.org/r/1690/ ........ Merged revisions 353454 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 00:00 +0000 [r353503] Terry Wilson * res/res_calendar.c, /: Allow res_calendar to be unloaded The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ ........ Merged revisions 353502 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 15:05 +0000 [r353551] Matthew Jordan * /, contrib/init.d/etc_default_asterisk: Added clarification for the VERBOSITY setting to etc_default_asterisk Clarified that using the VERBOSITY setting in etc_default_asterisk is the same as using the -v command line switch, which causes Asterisk to launch in console mode. (closes issue ASTERISK-17030) Reported by: Jonas ........ Merged revisions 353550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 15:51 +0000 [r353599] Sean Bright * /, include/asterisk/audiohook.h: Resolve an overlap in the ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused unintended side effects. This patch moves AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention. This will affect existing modules that use these flags, so be sure to recompile as necessary. (closes issue ASTERISK-19246) Reported by: feyfre ........ Merged revisions 353598 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 21:16 +0000 [r353771-353721] Jonathan Rose * /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt available which are slightly more readable than using a direct call to ast_sockaddr_stringify_fmt. This patch switches a number of those calls in chan_sip to use those wrappers and is generally harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. Young Patches: chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 353720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix sip show peers port output, align columns, and fix ami port output. A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for the AMI action "sippeers" which this patch changes back. Also, this aligns the output for the cli command "sip show peers" and fixes another issue that patch introduced by using ast_sockaddr_stringify calls multiple times without immediately using the pointer. I also went ahead and did a little janitorial work to clean up whitespace in _sip_show_peers. (issue ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by Walter Doekes (license 5674) ........ Merged revisions 353769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-02 18:48 +0000 [r353820] Mark Michelson * configs/http.conf.sample, main/manager.c, /, main/http.c, configs/manager.conf.sample, include/asterisk/manager.h: Fix TLS port binding behavior as well as reload behavior: * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 ........ Merged revisions 353770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-02 20:11 +0000 [r353868] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888) This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 353867 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-02 22:27 +0000 [r353916] Kinsey Moore * /, channels/chan_sip.c: Ensure entering T.38 passthrough does not cause an infinite loop After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban ........ Merged revisions 353915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-03 16:22 +0000 [r354000-353962] Jonathan Rose * res/res_fax.c: Fixes a segfault occuring when performing attended transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184) Reported by: Alexandr * /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to r335976 Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ ........ Merged revisions 353999 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-06 17:31 +0000 [r354217-354119] Richard Mudgett * /, main/features.c: Add missing headers to AMI UnParkedCall event to uniquely identify the call. The AMI UnParkedCall event was missing the Parkinglot and Uniqueid headers that the AMI ParkedCall event contains. (closes issue ASTERISK-19240) Reported by: Michael Yara ........ Merged revisions 354116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add extension" command. * Documented dialplan add extension ,,)> format. * Allow acceptance of command without the app-data value. There are many applications that do no need any parameters so it is silly to require that field for all commands. * Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2() calls. (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested by: rmudgett ........ Merged revisions 354216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-07 15:19 +0000 [r354270] Jonathan Rose * /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload for cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep its current data and then add a second copy during the reload. This would cause attempts to log the CDR to the database to fail. This patch also cleans up some unnecessary null checks for ast_free and deals with a few potential locking problems. (closes issue ASTERISK-19216) Reported by: Jacek Konieczny Review: https://reviewboard.asterisk.org/r/1711/ ........ Merged revisions 354263 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-07 21:17 +0000 [r354349] Terry Wilson * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql: Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing instead of "" 2. Don't set ipaddr or port to the string "(null)" when they are empty 3. Add missing required fields, set default for lastms to 0, and modify the length of the ipaddr field to 45 in the Postgresql realtime.sql file. (closes issue ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ ........ Merged revisions 354348 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 02:25 +0000 [r354493] Russell Bryant * main/channel.c, /: Remove some unnecessary locking from ast_hangup(). This patch removes some unnecessary locking of the channels container in ast_hangup(). The reason this came up is that this lock can very quickly block the entire system. If any of the channel cleanup code decides to block, it causes a problem for the whole system. For example, when audiohooks get destroyed, if that blocks for a while waiting on the mixmonitor thread to exit because it's busy blocking on some I/O, it causes a problem for many other threads in the meantime. Review: https://reviewboard.asterisk.org/r/1712/ ........ Merged revisions 354492 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 02:54 +0000 [r354496] Richard Mudgett * apps/app_parkandannounce.c, /: Fix crash in ParkAndAnnounce. Well, thats embarrasing. I forgot to initialize the caller_id storage. (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett ........ Merged revisions 354495 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 16:35 +0000 [r354543] Matthew Jordan * /, channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ Merged revisions 354542 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 17:04 +0000 [r354546] Mark Michelson * /, res/res_fax.c: Adding reload support to res_fax.so (closes issue ASTERISK-16712) reported by Frank DiGennaro Review: https://reviewboard.asterisk.org/r/1713 ........ Merged revisions 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 17:08 +0000 [r354548] Matthew Jordan * /, channels/chan_sip.c: Clean-up of minor formatting issues in r354542/3/4 rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ ........ Merged revisions 354547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 19:54 +0000 [r354703-354656] Kinsey Moore * /, main/config.c: Make the config parser remove escaping backslashes The config parser in Asterisk does not currently remove a backslash that is used to escape a semicolon which would otherwise be interpreted as the start of a comment. The change here causes that backslash to be removed, but does not create a real escape system in the config parser. The biggest complication with a real escape system would be breaking existing configs everywhere (parsing \\ as \ and breaking on escaped non-semicolon characters) even though it would be the "right" way to do things. (closes issue ASTERISK-17121) Review: https://reviewboard.asterisk.org/r/1724/ ........ Merged revisions 354655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix parsing of SIP headers where compact and non-compact headers are mixed Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ ........ Merged revisions 354702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 22:03 +0000 [r354750] Terry Wilson * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge is torn down CDRs cannot be modified after a bridge is torn down, (e.g. after Dial() returns) even though the CDR() function may be called. Since modifying the CDR code to change this behavior could very easily break all kinds of things, this patch just documents this limitation. (closes issues ASTERISK-16923) Review: https://reviewboard.asterisk.org/r/1720/ ........ Merged revisions 354749 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-10 18:05 +0000 [r354836] Richard Mudgett * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting to the same exten and context. The astman_get_header() never returns NULL so the check by the code for NULL would never fail. (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches: 0018325.patch (license #6116) patch uploaded by Nuno Borges (modified) ........ Merged revisions 354835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-10 22:00 +0000 [r354890] Jason Parker * apps/app_voicemail.c, /: Fix a voicemail memory leak with heard/deleted messages. open_mailbox() was changed quite a long time ago to allocate this memory. close_mailbox() should have been changed to be responsible for freeing it. ........ Merged revisions 354889 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-13 16:41 +0000 [r354938] Joshua Colp * apps/app_confbridge.c: Don't try to play sound files that do not exist. (closes issue ASTERISK-19188) Reported by: slesru 2012-02-13 17:24 +0000 [r354959] Richard Mudgett * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix reconnecting to pgsql database after connection loss. There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ ........ Merged revisions 354953 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-13 19:51 +0000 [r355010] Joshua Colp * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list. (closes issue AST-758) ........ Merged revisions 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-13 22:03 +0000 [r355057] Richard Mudgett * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file execution. Since the dir timestamp is available at one second resolution, we cannot know if it was updated within the same second after we scanned it. Therefore, we will force another scan if the dir was just modified. * Changed to force another scan if the directory was just modified. (closes issue ASTERISK-19081) Reported by: Knut Bakke Review: https://reviewboard.asterisk.org/r/1688/ ........ Merged revisions 355056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 09:49 +0000 [r355137] Alexandr Anikin * addons/chan_ooh323.c, /: call manager_event only if there is not null channel structure (Closes issue ASTERISK-19298) Reported by: robinfood Patches: issue19298.patch uploaded by may213 (License #5415) ........ Merged revisions 355136 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 13:33 +0000 [r355183] Sean Bright * /, channels/chan_iax2.c: Clear the high order bit from the destination call number before sending. send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. ........ Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 15:53 +0000 [r355229] Jason Parker * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs by default in sample configs. ........ Merged revisions 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 16:27 +0000 [r355271] Mark Michelson * /, channels/chan_sip.c: Properly invert the return of a strncmp call. This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) ........ Merged revisions 355268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 18:14 +0000 [r355375-355320] Richard Mudgett * /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported by: Alex Villacis Lasso Patches: asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch (license #5617) patch uploaded by Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1740/ ........ Merged revisions 355319 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configure, include/asterisk/autoconfig.h.in, configure.ac, formats/format_ogg_vorbis.c: Fix voicemail problems when using ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file format because it did not implement the seek and tell format callbacks among other problems. Since we were already using the libvorbis and libvorbisenc libraries we can use libvorbisfile as it is also part of the vorbis library package. * Made use the libvorbisfile to handle the ogg/vorbis file stream. The format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile. (closes issue ASTERISK-16926) Reported by: sque Patches: ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque ........ Merged revisions 355365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-15 17:25 +0000 [r355530-355449] Sean Bright * /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally intended. Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. ........ Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: Only use maxtrunkcall and maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These variables are only accessed from the IAX_OLD_FIND path, so there is no reason to keep them updated otherwise. ........ Merged revisions 355458 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make sure to log 'apathetic' messages too. ........ Merged revisions 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-16 18:32 +0000 [r355620-355575] Richard Mudgett * /, res/res_monitor.c: Fix AMI Monitor action without File header converting channel name into filename. * Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. ........ Merged revisions 355574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_c_declare_check.m4 (added), configure.ac, formats/format_ogg_vorbis.c: Fix compile problem when old version of libvorbisfile v1.1.2 is used. The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........ Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-16 20:01 +0000 [r355623] Sean Bright * /, main/audiohook.c: Revert a change to audio_audiohook_write_list that had no affect. When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-17 19:06 +0000 [r355733] Mark Michelson * /, channels/chan_sip.c: Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 ........ Merged revisions 355732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-17 19:34 +0000 [r355794-355747] Sean Bright * /, channels/chan_iax2.c: Pass the correct value to ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow trunkfreq to be greater than 1000ms. ........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-18 07:58 +0000 [r355851] Alec L Davis * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c, channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-18 17:02 +0000 [r355896-355895] Paul Belanger * /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 355839 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /: Revert commit 2012-02-19 17:50 +0000 [r355998-355902] Sean Bright * /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so that we can set it's port later. Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: Add some boilerplate documentation for IAXVAR and IAXPEER. ........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, /: Change some debug messages from LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back. ........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: Remove spurious warning when 'qualifyfreqnotok' is set successfully. (closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-21 04:30 +0000 [r356074] Kinsey Moore * main/ccss.c: Add missing newline to ccss state change notification Move along, nothing to see here... 2012-02-21 11:17 +0000 [r356108] Sean Bright * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-22 14:53 +0000 [r356215] Matthew Jordan * /, channels/chan_sip.c: Merged revisions 356214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is determined, the number of peers in the container can change, as the peers container is not locked. This could cause a buffer overrun when populating the array, if peers were added to the container after the array was created. Additionally, a memory leak of the allocated array would occur if a user caused the _show_peers method to return CLI_SHOWUSAGE. We now create a snapshot of the current peers using an ao2_callback with the OBJ_MULTIPLE flag. This size of the array is set to the number of peers that the iterator will iterate over; hence, if peers are added or removed from the peers container it will not affect the execution of the "sip show peers" command. Review: https://reviewboard.asterisk.org/r/1738/ (closes issue ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283) ........ 2012-02-22 21:18 +0000 [r356297] Terry Wilson * main/loader.c, res/res_calendar.c, /, include/asterisk/calendar.h: Track module use count for res_calendar If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ ........ Merged revisions 356291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-23 03:23 +0000 [r356431-356428] Paul Belanger * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337 ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ ........ r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines Add back strsep() function for previous commit ........ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) | 2 lines Missed one strsep() function ........ Merged revisions 356290,356335,356337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * addons/chan_ooh323.c, /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 356430 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-23 15:40 +0000 [r356476] Mark Michelson * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses. When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) (with some slight modifications prior to commit) ........ Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-23 19:52 +0000 [r356522] Richard Mudgett * /, channels/chan_sip.c, main/features.c: Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-24 15:07 +0000 [r356651-356605] Matthew Jordan * res/res_srtp.c, channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h, main/rtp_engine.c, /, include/asterisk/rtp_engine.h: Allow SRTP policies to be reloaded Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/res_srtp.c, /: Remove srtp_shutdown from res_srtp The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. ........ Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-24 18:27 +0000 [r356690] Richard Mudgett * /, channels/chan_sip.c, include/asterisk/tcptls.h, channels/sip/include/sip.h: Fix worker thread resource leak in SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-25 17:22 +0000 [r356798] Matthew Jordan * apps/app_voicemail.c, /: Fix crash in app_voicemail during close_mailbox In r354890, a memory leak in app_voicemail was fixed by properly disposing of the allocated heard/deleted pointers. However, there are situations, particularly when no messages are found in a folder, where these pointers are not allocated and not NULL. In that case, an invalid free would be attempted, which could crash app_voicemail. As there are a number of code paths where this could occur, this patch uses the number of messages detected in the folder before it attempts to free the pointers. This resolves the crash detected in the Asterisk Test Suite's check_voicemail_nominal test. ........ Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-27 15:30 +0000 [r356961] Jonathan Rose * /, res/res_odbc.c: Remove possible segfaults from res_odbc by adding locks around usage of odbc handle (closes issue ASTERISK-19011) Reported by: Walter Doekes Patches: issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674) review: https://reviewboard.asterisk.org/r/1719/ review: https://reviewboard.asterisk.org/r/1622/ ........ Merged revisions 356917 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-27 16:05 +0000 [r356964] Terry Wilson * /, main/features.c: Copy CDR variables when set during a bridge This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ ........ Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-27 23:36 +0000 [r357095] Richard Mudgett * main/channel.c, /: Fix callerid of Originated calls. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix ........ Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 Asterisk Development Team * Asterisk 10.2.0 Released. 2012-03-01 Asterisk Development Team * Asterisk 10.2.0-rc4 Released. * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0. In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. 2012-02-28 Asterisk Development Team * Asterisk 10.2.0-rc3 Released. * main/channel.c: Fix callerid of Originated calls. The callerid of originated calls (independent of mechanism) was not being passed to the outbound channel. This patch fixes that. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix * channels/chan_sip.c: Fix ACK routing for non-2xx responses. When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) * channels/chan_sip.c: Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 2012-02-10 Asterisk Development Team * Asterisk 10.2.0-rc2 Released. * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric codes. In ASTERISK-18924, SIP INFO DTMF handling was changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared against '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from uninitialized caller_id storage (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett * channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to r335976. Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ * channels/chan_sip.c: Ensure entering T.38 passthrough does not cause an infinite loop. After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban * channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. 2012-02-01 Asterisk Development Team * Asterisk 10.2.0-rc1 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1020RCS-2 2012-01-30 12:48 +0000 [r353261] Kevin P. Fleming * /, channels/chan_sip.c: Clarify log WARNING message when port-zero SDP 'm' lines received. Previously, if an m-line in an SDP offer or answer had a port number of zero, that line was skipped, and resulted in an 'Unsupported SDP media type...' warning message. This was misleading, as the media type was not unsupported, but was ignored because the m-line indicated that the media stream had been rejected (in an answer) or was not going to be used (in an offer). ........ Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-29 02:44 +0000 [r353176] Russell Bryant * main/netsock.c, /: Find even more network interfaces. The previous change made the code look for emN and pciN in addition to what it did originally, which was search for ethN. However, it needed to be looking for pciN#N, so that's what it does now. This also moves the memset() to be before every ioctl(). ........ Merged revisions 353175 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-28 14:51 +0000 [r353127] Kevin P. Fleming * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM) audio for quite some time, but some endpoints refer to it as 'L16-256'. This commit adds this as an alias for the existing format. ........ Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-28 04:27 +0000 [r353078] Russell Bryant * main/netsock.c, /: Update ast_set_default_eid() to find more network interfaces. As of Fedora 15, ethN is not the name of ethernet interfaces. The names are emN or pciN. Update some code that searched for interfaces named ethN to look for the new names, as well. For more information about why this change was made, see this page: http://domsch.com/blog/?p=455 ........ Merged revisions 353077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 21:37 +0000 [r352992-353039] Richard Mudgett * apps/app_queue.c: Audit of ao2_iterator_init() usage for v10. Missed one. * tests/test_format_api.c: Audit of ao2_iterator_init() usage for v10. Fix double format_cap iterator cleanup. 2012-01-27 19:19 +0000 [r352965] Jonathan Rose * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session. I also went ahead and took a little time to make sure that the manager value AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's how we handle this stuff these days. (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766) ........ Merged revisions 352959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 18:36 +0000 [r352956] Richard Mudgett * res/res_srtp.c, main/pbx.c, /, channels/chan_sip.c, include/asterisk/indications.h, res/snmp/agent.c, main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c, apps/app_chanspy.c, main/indications.c, res/res_odbc.c: Audit of ao2_iterator_init() usage for v1.8. Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 00:08 +0000 [r352863] Alec L Davis * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 352862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ ........ 2012-01-26 20:22 +0000 [r352817] Alexandr Anikin * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) ........ Merged revisions 352807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-26 19:07 +0000 [r352756] Jonathan Rose * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during create_addr_from_peer For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner ........ Merged revisions 352755 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-26 06:33 +0000 [r352705] Alec L Davis * /, channels/chan_sip.c: Merged revisions 352704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make similar to other Notify messages. sample output: terminated Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1693/ ........ 2012-01-25 22:23 +0000 [r352651] Paul Belanger * apps/app_voicemail.c, /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 352643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 21:18 +0000 [r352616] Kevin P. Fleming * /, main/test.c: Avoid unnecessary rebuilds of main/test.c. main/test.c includes "asterisk/version.h", when it should include "asterisk/ast_version.h" instead (and it should use the ast_get_version() and ast_get_version_num() functions). This commit modifies it to extract the Asterisk version information using the proper APIs, and as a result means that main/test.c no longer needs to be rebuilt when a Subversion checkout is updated or modified. ........ Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 17:30 +0000 [r352556] Terry Wilson * /, channels/chan_sip.c: Remove some extraneous debugging from registry memleak fix ........ Merged revisions 352551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 17:16 +0000 [r352520] Richard Mudgett * channels/chan_sip.c, CHANGES, main/message.c, channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ 2012-01-25 16:54 +0000 [r352516] Kevin P. Fleming * main/format.c, main/format_cap.c, main/format_pref.c: Eliminate unnecessary rebuilds of main/format*.c. These files have no need to include "asterisk/version.h", and doing so forces them to be rebuilt each time a Subversion checkout moves between 'modified' and 'unmodified' states. 2012-01-25 16:49 +0000 [r352515] Terry Wilson * /, channels/chan_sip.c: Clean up some SIP registry-related memory leaks 1) Be sure and free at unload the epa_backend we allocate at startup 2) Do the same sip_registry cleanup at unload we do at reload Review: https://reviewboard.asterisk.org/r/1689/ ........ Merged revisions 352514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 16:41 +0000 [r352512] Jonathan Rose * /, configs/sip.conf.sample: Redocuments sip types peer, user, friend in sip.conf.sample There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-24 22:22 +0000 [r352430] Mark Michelson * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured. (closes issue ASTERISK-16550) reported by: Olle Johansson ........ Merged revisions 352424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-24 20:35 +0000 [r352373] Jonathan Rose * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22! (closes issue ASTERISK-18978) Reported by: Cameron Twomey Patches: confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 uploaded by Cameron Twomey ........ Merged revisions 352367 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-24 17:02 +0000 [r352292] Richard Mudgett * /, funcs/func_odbc.c: Fix locking issues with channel datastores in func_odbc.c. * Fixed a potential memory leak when an existing datastore is manually destroyed by inline code instead of calling ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ ........ Merged revisions 352291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-23 20:30 +0000 [r352228-352231] Mark Michelson * /, main/features.c: Fix grammar of comment. ........ Merged revisions 352230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Fix blind transfers from failing if an 'h' extension is present. This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) Review: https://reviewboard.asterisk.org/r/1685 ........ Merged revisions 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-23 19:18 +0000 [r352149] Matthew Jordan * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) ........ Merged revisions 352144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-23 17:34 +0000 [r352091] Richard Mudgett * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the defined enum values. The invalid value used when notifycid was enabled was benign. As far as the code was concerned -1 and 1 are equivalent. (closes issue ASTERISK-19232) Reported by: Eike Kuiper ........ Merged revisions 352090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-21 00:21 +0000 [r352035] Richard Mudgett * /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time unit inconsistency. Note: Noone calls ast_app_dtget() with the timeout parameter of zero so the bad code normally will never get executed. * Fix unnecessary floating point division in func_timeout.c timeout_write() when all other values are integers. (closes issue ASTERISK-16817) Reported by: Dmitry Andrianov ........ Merged revisions 352029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-21 00:08 +0000 [r352015-352017] Mark Michelson * /, channels/chan_sip.c: Remove XXX comment that is not necessary. ........ Merged revisions 352016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference for the RTP instance of the transferer. This fixes the issue by merging two similar but slightly conflicting sections of code into a single area. It also adds a stop_media_flows() call in the case that the transferer's UA never sends a BYE to us like it is supposed to. (issue ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/ ........ Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-20 19:35 +0000 [r351816-351861] Kinsey Moore * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code These changes are in a file that is not compiled by default, and so were missed on earlier checks. ........ Merged revisions 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /: Recorded merge of revisions 351858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Allow ilbc code to build under dev mode GCC 4.6.3 found some set/unused variables in the ILBC code. * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore LSF_check function calls from set/unused variable removal These functions are not noops and modify the array that is passed in. Thanks for the catch Richard. * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove more set, but unused variables in the ilbc codec GCC 4.6.3 caught these in dev mode as well. 2012-01-20 15:59 +0000 [r351762] Jonathan Rose * /, channels/chan_sip.c: Adds setting of mwi_from field to check_auth_result check_peer_ok (closes ASTERISK-19057) Reported By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license 5242) ........ Merged revisions 351759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-20 15:54 +0000 [r351761] Matthew Jordan * codecs/ilbc/helpfun.c, /: Remove unused variable 'tmp' from helpfun in ilbc codec gcc version 4.6.2 caught an unused variable in the ilbc codec library. This would prevent compilation with --enable-dev-mode; variable removed. ........ Merged revisions 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-20 13:01 +0000 [r351708] Stefan Schmidt * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in the channels/sip folder like reqresp_parser.c ........ Merged revisions 351707 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 23:25 +0000 [r351646] Richard Mudgett * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in get_calleridname() parsing and ensure that the output buffer is nul terminated. * Make get_calleridname() truncate the name it parses if the given buffer is too small rather than abandoning the parse and not returning anything for the name. Adjusted get_calleridname_test() unit test to handle the truncation change. * Fix get_in_brackets_test() unit test to check the results of get_in_brackets() correctly. * Fix parse_name_andor_addr() to not return the address of a local buffer. This function is currently not used. * Fix potential NULL pointer dereference in sip_sendtext(). * No need to memset(calleridname) in check_user_full() or tmp_name in get_name_and_number() because get_calleridname() ensures that it is nul terminated. * Reply with an accurate response if get_msg_text() fails in receive_message(). This is academic in v1.8 because get_msg_text() can never fail. ........ Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 22:43 +0000 [r351612] Kinsey Moore * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter statistics in SR and RR reports Change the RTCP RR and SR generation code to convert Asterisk's internal jitter statistics to be represented in RTP timestamp units based on the rate of the codec in use instead of in seconds. (closes issue ASTERISK-14530) ........ Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 21:47 +0000 [r351560] Jonathan Rose * /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates doubling the :port part of SIP Notify Message-Account headers. This patch prevents the domain string from getting mangled during the initreqprep step by moving the initialization to before its immediate use. It also documents this pitfall for the ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ ........ Merged revisions 351559 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 21:12 +0000 [r351505] Joshua Colp * /, channels/chan_sip.c: Prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda ........ Merged revisions 351504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-18 21:05 +0000 [r351451] Matthew Jordan * codecs/ilbc/helpfun.c (added), codecs/ilbc/LICENSE_ADDENDUM (added), codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added), codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added), codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added), codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h (added), codecs/ilbc/iLBC_decode.h (added), codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added), codecs/ilbc/hpInput.c (added), codecs/ilbc/gainquant.c (added), codecs/ilbc/iCBSearch.h (added), codecs/ilbc/hpOutput.c (added), codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added), codecs/ilbc/hpInput.h (added), codecs/ilbc/LPCencode.c (added), codecs/ilbc/gainquant.h (added), codecs/codec_ilbc.c, codecs/ilbc/hpOutput.h (added), codecs/ilbc/StateSearchW.c (added), codecs/ilbc/PATENTS (added), contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LPCencode.h (added), codecs/ilbc/LICENSE (added), codecs/ilbc/StateSearchW.h (added), codecs/ilbc/iCBConstruct.c (added), codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h (added), codecs/ilbc/iLBC_test.c (added), codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c (added), codecs/ilbc/packing.c (added), codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c (added), codecs/ilbc/iLBC_encode.c (added), codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added), codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h (added), codecs/ilbc/extract-cfile.awk (added), codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile, codecs/ilbc/FrameClassify.h (added): Include iLBC source code for distribution with Asterisk This patch includes the iLBC source code for distribution with Asterisk. Clarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder. Review: https://reviewboard.asterisk.org/r/1675 Review: https://reviewboard.asterisk.org/r/1649 (closes issue: ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan ........ Merged revisions 351450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-18 15:57 +0000 [r351408] Stefan Schmidt * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't recognized a proper callerid name and number from a P-Asserted-Identity cause the header parsing logic was wrong. Changing the parsing functions to the sip header parsing APIs in reqresp_parser.h solves this problem. Review: https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and Mark Michelson ........ Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 17:22 +0000 [r351308] Mark Michelson * res/res_rtp_asterisk.c, /: Eliminate odd initialization of probation variable. ........ Merged revisions 351306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 17:08 +0000 [r351289] Jonathan Rose * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 16:54 +0000 [r351286] Mark Michelson * /, channels/chan_sip.c: Use built-in parsing functions for Contact and Record-Route headers. If a Contact or a Record-Route header had a quoted string with an item in angle brackets, then we would mis-parse it. For instance, "Bob <1234>" <1234@example.org> would be misparsed as having the URI "1234" The fix for this is to use parsing functions from reqresp_parser.h since they are heavily tested and are awesome. (issue ASTERISK-18990) ........ Merged revisions 351284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 16:07 +0000 [r351234] Matthew Jordan * /, channels/chan_sip.c: Fix udptl issue with initial INVITE introduced by r351027 When an inital INVITE occurs that contains image media, a channel is not yet associated with the SIP dialog. The file descriptor associated with the udptl session needs to be set in initialize_udptl or in sip_new to account for this scenario. ........ Merged revisions 351233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 01:43 +0000 [r351183] Russell Bryant * /, channels/chan_sip.c: Merged revisions 351182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines Add some missing locking in chan_sip. This patch adds some missing locking to the function send_provisional_keepalive_full(). This function is called from the scheduler, which is processed in the SIP monitor thread. The associated channel (or pbx) thread will also be using the same sip_pvt and ast_channel so locking must be used. The sip_pvt_lock_full() function is used to ensure proper locking order in a safe manner. In passing, document a suspected reference counting error in this function. The "fix" is left commented out because when the "fix" is present, crashes occur. My theory is that fixing it is exposing a reference counting error elsewhere, but I don't know where. (Or my analysis of this being a problem could have been completely wrong in the first place). Leave the comment in the code for so that someone may investigate it again in the future. Also add a bit of doxygen to transmit_provisional_response(). (closes issue ASTERISK-18979) Review: https://reviewboard.asterisk.org/r/1648 ........ 2012-01-16 21:17 +0000 [r351081-351131] Terry Wilson * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200 response to INVITE When handling a non-2xx final response on an INVITE transaction, we have to keep the transaction around after we send an ACK in case we receive a retransmission of the response so we can re-transmit the ACK, but also tear down the ast_channel as soon as we transmit the ACK. Before this patch, we could fail at both of these things. Calling sip_alreadygone/needdestroy prevented us from keeping the transaction up and retransmitting the ACK, and queueing CONGESTION was not sufficient to cause the channel to be torn down when originating calls via the CLI, for example. This patch queues a hangup with CONGESTION instead of just queueing CONGESTION for these responses and removes the sip_alreadygone and sip_needdestroy calls from handle_response_invite on non-2xx responses. It relies on the hangup calling sip_scheddestroy. For more information, see section 17.1.1.1 of RFC 3261. (closes issue ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/ ........ Merged revisions 351130 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Don't prematurely stop SIP session timer When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry. (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: session_timer_fix.diff by Terry Wilson (License #5357) based on session_timer.patch by Thomas Arimont (License #5525) ........ Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-16 19:12 +0000 [r351028] Matthew Jordan * /, channels/chan_sip.c: Create and initialize udptl only when dialog negotiates for image media Prior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or what an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This occurred even in non-INVITE dialogs that would never send image media. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates through the appropriate control frame that a dialog is to support T.38. (closes issue ASTERISK-16698) Reported by: under Tested by: Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar Broad Tested by: Stefan Schmidt review: https://reviewboard.asterisk.org/r/1668/ ........ Merged revisions 351027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-16 17:11 +0000 [r350978] Sean Bright * main/db.c: Sort the output of 'database showkey' as well. You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. 2012-01-16 17:06 +0000 [r350976] Joshua Colp * main/rtp_engine.c, /: Add missing code to set direct RTP setup information during dialing. ........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-16 14:27 +0000 [r350938] Sean Bright * main/db.c: Sort the output of 'database show' by key. This more closely mimics the behavior of 'database show' before the conversion to sqlite3. 2012-01-15 20:12 +0000 [r350886-350889] Walter Doekes * /, main/asterisk.c: Allow only one thread at a time to do asterisk cleanup/shutdown. Add locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1662/ Review: https://reviewboard.asterisk.org/r/1658/ ........ Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile error in utils/extconf.c. Note that I'm not confirming legitimacy of having that file in tree at all. Is anyone using aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged revisions 350885 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-14 16:41 +0000 [r350790-350838] Kevin P. Fleming * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac, autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the configure script are properly quoted. Recent versions of autoconf (2.68 on my system) won't properly process the configure script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script were, but many were not. This patch corrects the unquoted calls. ........ Merged revisions 350837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/scripts/install_prereq, /, channels/chan_h323.c, addons/chan_mobile.c, res/res_pktccops.c: Multiple revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites are properly installed on Debian-style distributions. * Don't specify a specific version of libgmime; newer versions are available now and acceptable. * Install libsrtp so that res_srtp can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines Correct some 'set-but-not-used' variable warnings. ........ Merged revisions 350788-350789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 22:10 +0000 [r350737] Kinsey Moore * /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for the ASTERISK-18929 fix configure and autoconfig.h.in were not regenerated when the fix was committed. ........ Merged revisions 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 21:51 +0000 [r350734] Richard Mudgett * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample: Correct eventtype names in cel_odbc and cel_pgsql sample files ........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 21:41 +0000 [r350731] Kinsey Moore * /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller ........ Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 20:31 +0000 [r350680] Mark Michelson * /, channels/sip/config_parser.c: Set port to a default sane value if a bogus one is provided when parsing hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 17:29 +0000 [r350585] Richard Mudgett * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c, configs/cel.conf.sample, /, cel/cel_manager.c, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample, main/cel.c, configs/cel_custom.conf.sample: Add missing CEL logging fields to various CEL backends. Multiple revisions 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) ........ Merged revisions 350555,350571 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 16:59 +0000 [r350550-350553] Matthew Jordan * /, apps/app_queue.c: Realtime queues failed to load queue information without queue member table Previously, realtime queues could be loaded without defining the queue member table. This allowed for queue members to be dynamic, while the realtime queue definitions could exist in some backing storage. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. Previously, an empty ast_config object was expected. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * bridges/bridge_builtin_features.c, channels/chan_bridge.c, include/asterisk/bridging.h, apps/app_confbridge.c, main/bridging.c: Fix crash from bridge channel hangup race condition in ConfBridge This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ 2012-01-12 16:04 +0000 [r350502] Jonathan Rose * /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) ........ Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-11 22:51 +0000 [r350312-350453] Richard Mudgett * /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel. (closes issue ASTERISK-19180) Reported by: Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK function. The time passed by the LOCK function to an internal function was relative time when the function expected absolute time. * Don't use C++ keywords in get_lock(). (closes issue ASTERISK-16868) Reported by: Andrey Solovyev Patches: 20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified) ........ Merged revisions 350311 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-09 21:55 +0000 [r350221] Richard Mudgett * /, channels/chan_iax2.c: Fix joinable thread terminating without joiner memory leak in chan_iax.c. The iax2_process_thread() can exit without anyone waiting to join the thread. If noone is waiting to join the thread then a large memory leak occurs. * Made iax2_process_thread() deatach itself if nobody is waiting to join the thread. (closes issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified) (closes issue ASTERISK-17825) Reported by: wangjin ........ Merged revisions 350220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-09 19:34 +0000 [r350180] Walter Doekes * main/db.c: Fix shutdown handling of sqlite3 astdb. If a db_sync was scheduled just before shutdown, the atexit code calling db_sync would have no effect, causing the astdb commit thread to stay alive. This caused the SIP/realtime_sipregs test to fail. (The fallback kill would run the atexit code again and that would wreak havoc.) This fixes that the atexit kill condition is picked up properly. (closes issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1659 2012-01-09 18:57 +0000 [r350076-350129] Richard Mudgett * /, contrib/scripts/valgrind_compare (added): Multiple revisions 350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script live_ast to invoke Asterisk with valgrind and suppression file. * Added valgrind_compare script to compare two valgrind log files for differences. (issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: valgrind_compare (license #5035) script uploaded by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger ........ r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 350127-350128 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/asterisk.c: Make Asterisk -x command line parameter imply -r parameter presence. The Asterisk -x command line parameter is documented inconsistently. * Made the -x documentation and behavior consistent. * Since this is also a new year, updated the copyright notices while here. (closes issue ASTERISK-19094) Reported by: Eugene Patches: issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified) Tested by: Eugene ........ Merged revisions 350075 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-09 15:39 +0000 [r350024] Kinsey Moore * /, apps/app_meetme.c: Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-06 23:25 +0000 [r349977] Terry Wilson * /, channels/chan_sip.c: Don't leak CID in From header when presentation=unavailable When someone does Set(CALLERPRES()=unavailable) (or Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows "Anonymous" . When sendrpid=yes/pai, the From header will still display the callerid info, even though we supply an rpid header with the anonymous info. It seems like we shouldn't leak that info in any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems to indicate that one shouldn't send identifying info in the From in this case. This patch anonymizes the From header as well even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review: https://reviewboard.asterisk.org/r/1649/ ........ Merged revisions 349968 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-06 21:25 +0000 [r349928] Kinsey Moore * pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected behavior with confbridge A bug in the pbx_lua goto detection was causing the dialplan to hangup unexpectedly after confbridge exited if it had called lua dialplan code during execution. Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue ASTERISK-18976) 2012-01-06 16:48 +0000 [r349873] Richard Mudgett * /, apps/app_followme.c: Fix memory leaks in app_followme find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt Jordan ........ Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-05 23:56 +0000 [r349822] Matthew Jordan * res/res_fax.c: Fix premature free'ing of the frame committed in r349608 Even though we set the frame to the ast_null_frame and return that, the caller of the frame hook may still need the frame. This now is a bit more careful about when it frees the frame, i.e., only under the same conditions that applied when we duplicated it in the first place. 2012-01-05 23:46 +0000 [r349820] Richard Mudgett * /, cel/cel_sqlite3_custom.c: Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId. If a table is created by some other application and the primary key is not named "AcctId", cel/cel_sqlite3_custom.c will always try to create the table and fail because it already exists. * Change the SQL table query to not require AcctId as the primary key. (closes issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch (license #6337) patch uploaded by socketpair ........ Merged revisions 349819 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-05 22:10 +0000 [r349732] Kinsey Moore * /, main/file.c: Allow playback of formats that don't support seeking ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-05 21:55 +0000 [r349673-349729] Jonathan Rose * main/dsp.c, /: Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ ........ Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/asterisk.c: Ensures Asterisk closes when receiving terminal signals in 'no fork' mode. When catching a signal, in no fork mode the console thread is identical to the thread responsible for catching the signal and closing Asterisk, which requires it to first dispense with the console thread. Prior to this patch, if these threads were identical, upon receiving a killing signal, the thread will send an URG signal to itself, which we also catch and then promptly do nothing with. Obviously this isn't useful behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-04 22:19 +0000 [r349608-349619] Matthew Jordan * apps/confbridge/conf_config_parser.c: Fix for ConfBridge config parser unlocking channel mutex too many times When looking up a ConfBridge profile, the config parser would, if it found a channel datastore on the channel requesting the bridge profile, unlock the channel mutex twice. Since that's a little aggressive, it now only unlocks it once. (closes issue ASTERISK-19042) Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042 uploaded by David Vossel (license 5628) * res/res_fax.c: Free successfully translated frame in fax_gateway_framehook A frame that is translated via ast_translate is also duplicated via ast_frdup. This will allocate a new frame on the heap, which needs to be free'd at the appropriate time. This issue reporter used valgrind to find that this occurred in res_fax's fax_gateway_framehook; a quick search through the code showed that only place this was currently not handling the translatted frame properly. (closes issue ASTERISK-19133) Reported by: Sylvain Rochet 2012-01-04 20:50 +0000 [r349559] Richard Mudgett * channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private pointer checks in the following chan_dahdi channel callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by: Diego Aguirre Tested by: rmudgett ........ Merged revisions 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-04 20:23 +0000 [r349505-349532] Kinsey Moore * contrib/init.d/rc.debian.asterisk, /: Make debian init script conform to the LSB standard Previously, this init script would return 1 if Asterisk was already running. This is incorrect behavior according to the LSB standard and has been fixed by returning 0 instead. (closes issue ASTERISK-17958) Reported-by: johnc ........ Merged revisions 349529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: Update autosupport script and man page Added information collection from the output of the utilities: top, free, uptime, ifconfig Added information collection from the output of the Asterisk command 'dahdi show status' Added option / flag '-n, --non-interactive' Updated man page to reflect new option / flag '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes issue AST-749) ........ Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-04 19:44 +0000 [r349451-349502] Jonathan Rose * /, channels/chan_sip.c: Adds Subscription-State header to notify with call completion. per RFC3265 (Closes issue ASTERISK-17953) Reported by: George Konopacki Patches: 19400.patch uploaded by mmichelson (license 5049) ........ Merged revisions 349482 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/pbx.c, /: Fix documentation for SayNumber to reflect the fact that language is changed in CHANNEL() (closes issue ASTERISK-18962) reported by: Nir Simionovich ........ Merged revisions 349450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 Asterisk Development Team * Asterisk 10.1.0 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-4 2012-01-24 Asterisk Development Team * Asterisk 10.1.0-rc2 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-3 * main/file.c: Allow playback of formats that don't support seeking. ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. (closes issue ASTERISK-18994) Patch-by: Timo Teras * channels/chan_sip.c: AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda * channels/chan_sip.c: Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferer. (closes issue ASERISK-19192) Reported by: Tyuta Vitali * res/res_rtp_asterisk: Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is the default setting in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. * main/features.c: Fix blind transfers from failing if an 'h' extension is present. This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) * apps/app_queue.c: Realtime queues failed to load queue information without queue member table. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza 2011-12-30 Asterisk Development Team * Asterisk 10.1.0-rc1 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-1 2011-12-29 15:14 +0000 [r349340] Matthew Jordan * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (issue ASTERISK-19040) (issue ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/ ........ Merged revisions 349339 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-28 21:33 +0000 [r349290] Sean Bright * /, main/audiohook.c: Use ast_audiohook_write_list_empty to determine if our lists are empty instead of duplicating that logic. ........ Merged revisions 349289 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-28 19:00 +0000 [r349248-349250] Kevin P. Fleming * utils: Tell Subversion to gnore the 'astdb2bdb' binary file if it exists. * main/dsp.c, res/res_fax.c, include/asterisk/dsp.h, include/asterisk/res_fax.h, res/res_fax_spandsp.c: Improve T.38 gateway V.21 preamble detection. This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). 2011-12-27 20:53 +0000 [r349195] Matthew Jordan * /, res/res_timing_pthread.c, include/asterisk/module.h, res/res_timing_dahdi.c, res/res_timing_timerfd.c, res/res_musiconhold.c: Fix timing source dependency issues with MOH Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ ........ Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-27 17:17 +0000 [r349145] Sean Bright * /, main/audiohook.c: Once an audiohook is attached to a channel, we continue to transcode all of the frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. ........ Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-23 17:32 +0000 [r349045] Sean Bright * /, apps/app_chanspy.c: Merged revisions 349044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines In ChanSpy, don't create audiohooks that will never be used. When ChanSpy is initialized it creates and attaches 3 audiohooks: 1) Read audio off of the channel that we are spying on 2) Write audio to the channel that we are spying on 3) Write audio to the channel that is bridged to the channel that we are spying on. The first is always necessary, but the others are used only when specific options are passed to the ChanSpy application (B, d, w, and W to be specific). When those flags are not passed, neither of those audiohooks are ever sent frames, but we still try to process the hooks for each voice frame that we recieve on the channel. So in short - only create and attach audiohooks that we actually need. ........ 2011-12-23 15:25 +0000 [r348993] Kinsey Moore * apps/app_dial.c, /: Fix missing doc tags found while fixing ASTERISK-18689 Add missing tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-23 02:30 +0000 [r348952] Richard Mudgett * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix extension state callback references in chan_sip. Chan_sip gives a dialog reference to the extension state callback and assumes that when ast_extension_state_del() returns, the callback cannot happen anymore. Chan_sip then reduces the dialog reference count associated with the callback. Recent changes (ASTERISK-17760) have resulted in the potential for the callback to happen after ast_extension_state_del() has returned. For chan_sip, this could be very bad because the dialog pointer could have already been destroyed. * Added ast_extension_state_add_destroy() so chan_sip can account for the sip_pvt reference given to the extension state callback when the extension state callback is deleted. * Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy() and handle_statechange() now that the struct ast_state_cb has a destructor to call. * Ensure that ast_extension_state_add_destroy() will never return -1 or 0 for a successful registration. * Fixed pbx.c statecbs_cmp() to compare the correct information. The passed in value to compare is a change_cb function pointer not an object pointer. * Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for deadlocking when those locks are held during the callback. * Removed unused lock declaration for the pbx.c store_hints list. (closes issue ASTERISK-18844) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/1635/ ........ Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-22 22:37 +0000 [r348846-348889] Matthew Jordan * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql There were a number of issues in cel_pgsql's pgsql_log method: * If either sql or sql2 could not be allocated, the method would return while the pgsql_lock was still locked * If the execution of the log statement succeeded, the sql and sql2 structs were never free'd * Reconnection successes were logged as ERRORs. In general, the severity of several logging statements was reduced (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/ ........ Merged revisions 348888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/say.c, main/file.c, main/app.c, apps/app_confbridge.c, main/bridging.c: Add Asterisk TestSuite event hooks to support ConfBridge testing This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) 2011-12-22 20:17 +0000 [r348845] Terry Wilson * /, include/asterisk/format_pref.h: Allow packetization vaules > 127 According to the RTP packetization documentation, and the maximum values listed in AST_FORMAT_LIST, we should support values > that the signed char array that ast_codec_pref makes available to store the value. All places in the code treat the framing field as though it were an int array instaead of a char array anyway, so this just fixes the type of the array. (closes issue ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/ ........ Merged revisions 348833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-21 20:13 +0000 [r348736-348793] Richard Mudgett * codecs/speex: Make codecs/speex ignore *.i files also. * apps/confbridge: Make apps/confbridge ignore *.i files also. * /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number if it is blank. Some ISDN switches complain or block the call if the RDNIS number is empty. * Made chan_iax2 not save a RDNIS number into the ast_channel if the string is blank. This is what other channel drivers do. (closes issue ASTERISK-17152) Reported by: rmudgett ........ Merged revisions 348735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-19 21:37 +0000 [r348648] Richard Mudgett * /, configure, configure.ac: Fix crashes on other platforms caused by interference from Darwin weak symbol support. Support weak symbols on a platform specific basis. The Mac OS X (Darwin) support must be isolated from the other platforms because it has caused other platforms to crash. Several other platforms including Linux have GCC versions that define the weak attribute. However, this attribute is only setup for use in the code by Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged revisions 348647 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-19 19:48 +0000 [r348605] Leif Madsen * main/message.c: Update documentation for MESSAGE_SEND_STATUS variable. (Closes issue ASTERISK-19056) Reported by: Yuri Patches: 348360.diff uploaded by Yuri (license #5242) 2011-12-18 18:28 +0000 [r348517] Kevin P. Fleming * /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample related to AST-2011-013. * The sample file listed *two* values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-16 23:56 +0000 [r348311-348465] Richard Mudgett * main/channel.c, /, main/features.c: Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). * Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. ........ Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, /: Fix cut and past error in ast_call_forward(). (issue ASTERISK-18836) ........ Merged revisions 348401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, main/pbx.c, /, apps/app_authenticate.c, funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix crash during CDR update. The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_parkandannounce.c, /: Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. ........ Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-14 22:34 +0000 [r348265] Matthew Jordan * apps/app_originate.c: Added support for all slin formats to app_originate Previously, app_originate could not originate a call into a non-8kHz conference bridge as the formats for non-8kHz slin codecs were not applied to the created channel. This patch adds all of the formats by default, such that if a created channel has a codec that supports a higher sampling rate, a translation path can be built between it and other channels. 2011-12-14 22:05 +0000 [r348213] Matthew Nicholson * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or receiving. The user may set that variable. ASTERISK-18921 ........ Merged revisions 348212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-14 21:58 +0000 [r348211] Matthew Jordan * apps/app_queue.c: Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has two required parameters (queuename, option). It was only checking for the presence of queuename. The patch checks for the existence of the option parameter and provides better error logging when invalid values are provided for the option parameter as well. 2011-12-14 20:35 +0000 [r348155-348158] Jonathan Rose * /, configs/features.conf.sample: Fix accidental use of tabs instead of spaces from previous features.conf.sample change ........ Merged revisions 348157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configs/features.conf.sample: Document PARKINGSLOT variable in features.conf.sample (issue ASTERISK-16239) ........ Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-13 23:06 +0000 [r348102] Richard Mudgett * /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix FollowMe CallerID on outgoing calls. The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-13 15:20 +0000 [r348056] Stefan Schmidt * channels/chan_sip.c: Fix possible misshandling of an incoming SIP response as a peer poke response. Also make sure peer has even qualify enabled when handle a peer poke response. (closes issue ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed by: David Vossel ........ Merged revisions 348048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-12 19:24 +0000 [r347996] Terry Wilson * res/res_srtp.c, /: Add a separate buffer for SRTCP packets The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) ........ Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-12 18:13 +0000 [r347953-347955] Richard Mudgett * configs/extensions.conf.sample, configs/iax.conf.sample, configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample, configs/vpb.conf.sample, configs/extensions.lua.sample, configs/sip.conf.sample: Reverting -r347953 for ASTERISK-14122 * configs/extensions.conf.sample, configs/iax.conf.sample, configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample, configs/vpb.conf.sample, configs/extensions.lua.sample, configs/sip.conf.sample: Update sample configs to put incoming calls into context public. * Add warning about the SIP allowguest option in context public. (closes issue ASTERISK-14122) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/ 2011-12-09 01:29 +0000 [r347812] Richard Mudgett * main/pbx.c, /: Fix some parsing issues in add_exten_to_pattern_tree(). * Simplify compare_char() and avoid potential sign extension issue. * Fix infinite loop in add_exten_to_pattern_tree() handling of character set escape handling. * Added buffer overflow checks in add_exten_to_pattern_tree() character set collection. * Made ignore empty character sets. * Added escape character handling to end-of-range character in character sets. This has a slight change in behavior if the end-of-range character is an escape character. You must now escape it. * Fix potential sign extension issue when expanding character set ranges. * Made remove duplicated characters from character sets. The duplicate characters lower extension matching priority and prevent duplicate extension detection. * Fix escape character handling when the escape character is trying to escape the end-of-string. We could have continued processing characters after the end of the exten string. We could have added the previous character to the pattern matching tree incorrectly. (closes issue ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-08 21:31 +0000 [r347727] Walter Doekes * /, channels/chan_sip.c: Fix regression when using tcpenable=no and tlsenable=yes. The tlsenable settings are tucked away in main/tcptls.c, so I missed them when resolving ASTERISK-18837. This should resolve the test suite breakage of the sip tls tests. Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt Jordan ........ Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-08 20:43 +0000 [r347656] Jonathan Rose * apps/app_queue.c: Fix regressed behavior of queue set penalty to work without specifying 'in ' r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ 2011-12-08 17:53 +0000 [r347600] Richard Mudgett * /, main/features.c: Mark channel running the h exten with the soft-hangup flag. When a bridge is broken, ast_bridge_call() might execute the h exten on the calling channel. However, that channel may not have been the channel that broke the bridge by hanging up. The channel executing the h exten must be in a hung up state so things like AGI run in the correct mode. * Make sure ast_bridge_call() marks the channel it is executing the h exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to match the pbx.c main dialplan execution loop when it executes the h exten.) (closes issue ASTERISK-18811) Reported by: David Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: David Hajek, rmudgett ........ Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-08 16:20 +0000 [r347532] Terry Wilson * /, channels/chan_sip.c: Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) ........ Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-07 21:39 +0000 [r347439] Richard Mudgett * main/manager.c, /: Update AMI Getvar and Setvar documentation about supplying a channel name. (closes issue ASTERISK-18958) Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-07 20:27 +0000 [r347383] Jonathan Rose * /, apps/app_meetme.c: Fix: Meetme recording variables from realtime DB use null entries over channel variables Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-07 20:12 +0000 [r347344] Terry Wilson * Makefile, include/asterisk/paths.h, configs/asterisk.conf.sample, build_tools/make_defaults_h, main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of configurable paths This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ 2011-12-06 23:56 +0000 [r347293] Richard Mudgett * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay signals case insensitive. (closes issue ASTERISK-18924) Reported by: Kevin Taylor ........ Merged revisions 347292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-06 21:53 +0000 [r347240] Jonathan Rose * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten If waitExten specifies a music class to use with its music on hold option, it will use CHANNEL(musicclass) instead if that channel variable has been set on the initiating channel. This documents that behavior in the waitExten app so that this can be known without checking the documentation of the code in function local_ast_moh_start. (closes issue ASTERISK-18804) ........ Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-06 19:42 +0000 [r347124-347167] Walter Doekes * /, channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no. When tcpenable=no, sending to transport=tcp hosts was still allowed. Resolving the source address wasn't possible and yielded the string "(null)" in SIP messages. Fixed that and a couple of not-so-correct log messages. (closes issue ASTERISK-18837) Reported by: Andreas Topp Review: https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan ........ Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Add regression tests for issue ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Move setting of voicemail zonetag and locale up a bit. The voicemail [general] zonetag and locale variables weren't loaded until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-06 17:24 +0000 [r347068] Matthew Jordan * /, channels/chan_sip.c: Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-05 17:42 +0000 [r347007] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. * Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patches: chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.h.diff (license #5685) patch uploaded by Richard Miller ........ Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-05 15:02 +0000 [r346955] Jonathan Rose * main/pbx.c, /: Resolve duplicate label used in multiple priorities for the same extension. Prior to this patch, if labels with the same name were used for different priorities in the same extension, the new label would be accepted, but it would be unusable since attempts to reach that label would just go to the first one. Now pbx.c detects this, generates a warning in logs, and culls the label before adding it to the dialplan. (closes issue ASTERISK-18807) Reported by: Kenneth Shumard Patches: pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........ Merged revisions 346954 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-05 14:46 +0000 [r346952] Kinsey Moore * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load regression introduced in r346087 Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-04 10:03 +0000 [r346900] Walter Doekes * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and domain ACL bypass. The code that allowed admins to create users with domain-only uri's had stopped to work in 1.8 because of the reqresp parser rewrites. This is fixed now: if you have a [mydomain.com] sip user, you can register with useraddr sip:mydomain.com. Note that in that case -- if you're using domain ACLs (a configured domain list) -- mydomain.com must be in the allow list as well. Reviewboard r1606 shows a list of registration combinations and which SIP response codes are returned. Review: https://reviewboard.asterisk.org/r/1533/ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes issue ASTERISK-18741) ........ Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-02 23:27 +0000 [r346856] Matthew Jordan * channels/chan_sip.c: Update SIP MESSAGE To parsing to correctly handle URI The previous patch (r346040) incorrectly parsed the URI in the presence of a port, e.g., user@hostname:port would fail as the port would be double appended to the SIP message. This patch uses the parse_uri function to correctly parse the URI into its username and hostname parts, and places them in the correct fields in the sip_pvt structure. (issue ASTERISK-18903) Review: https://reviewboard.asterisk.org/r/1597/ 2011-12-02 16:42 +0000 [r346763] Alexandr Anikin * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions 346762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under ........ 2011-12-01 21:14 +0000 [r346701] Richard Mudgett * main/stun.c, /, res/res_stun_monitor.c, configs/res_stun_monitor.conf.sample, include/asterisk/stun.h: Re-resolve the STUN address if a STUN poll fails for res_stun_monitor. The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ ........ Merged revisions 346700 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-01 20:37 +0000 [r346565-346698] Jonathan Rose * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing. 183 is actually a session progress message. (closes issue ASTERISK-18925) Reported by: Sebastian Denz Tested by: jrose Patches: asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139) ........ Merged revisions 346697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-30 19:37 +0000 [r346473] Leif Madsen * /, configs/queues.conf.sample: Update queues.conf.sample documentation. Update the documentation surrounding the use of MONITOR_EXEC to make it more clear that it can be used for both Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413) Reported by: David Woolley Patches: issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026) ........ Merged revisions 346472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-29 00:00 +0000 [r346349] David Vossel * include/asterisk/message.h, main/message.c: Fixes memory leak in message API. The ast_msg_get_var function did not properly decrement the ref count of the var it retrieves. The way this is implemented is a bit tricky, as we must decrement the var and then return the var's value. As long as the documentation for the function is followed, this will not result in a dangling pointer as the ast_msg structure owns its own reference to the var while it exists in the var container. 2011-11-28 14:32 +0000 [r346293] Stefan Schmidt * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified. (closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 22:58 +0000 [r346240] Richard Mudgett * include/asterisk/acl.h, /, channels/chan_skinny.c, channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls to ast_get_ip() not initializing the address family. ........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 20:45 +0000 [r346145-346198] Walter Doekes * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text() function. In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, include/asterisk/strings.h: Fix ast_str_truncate signedness warning and documentation. Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 17:14 +0000 [r346087] Kinsey Moore * channels/chan_jingle.c, /, include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource leaks This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 16:19 +0000 [r346040] Matthew Jordan * channels/chan_sip.c: Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ 2011-11-23 16:10 +0000 [r346031] Terry Wilson * /, res/res_musiconhold.c: Resume playing existing hold music for cached realtime MOH As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 16:06 +0000 [r346029] Paul Belanger * res/res_format_attr_celt.c, res/res_format_attr_silk.c: Added support level for new modules 2011-11-22 23:00 +0000 [r345977] Richard Mudgett * main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries to ask for the same address family each time. The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-22 20:31 +0000 [r345924] Walter Doekes * include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros exist next to the LOG_* macros. (issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-22 16:40 +0000 [r345882] Paul Belanger * apps/confbridge/conf_config_parser.c: Add missing sound_only_one config variable (closes issue ASTERISK-18895) Reported by: zvision Patches: conf_config_parser.diff (license #5755) patch uploaded by zvision 2011-11-21 21:07 +0000 [r345830] Terry Wilson * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default to nat=yes; warn when nat in general and peer differ It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-19 15:10 +0000 [r345640-345683] Tilghman Lesher * /, main/db.c: Update the documentation to better clarify how the existing commands work. Review: https://reviewboard.asterisk.org/r/1593/ ........ Merged revisions 345682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/db.c: Fix a change in behavior in 'database show' from 1.8. In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev 2011-11-17 17:29 +0000 [r345558] Richard Mudgett * /, channels/sig_pri.c: Remove dead code since pri_grab() can never fail. Dead code makes programmers sick. I am sick of looking at it. ........ Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-16 14:49 +0000 [r345488] Jonathan Rose * apps/app_voicemail.c, /: Guarantee messages go into the right folders with multiple recipients Before, using the U flag in Voicemail with multiple recipients would put urgent messages in the INBOX folder for all users past the first thanks to a bug with the message copying function. This would also cause messages to fail to be sent if the INBOX directory hadn't been created for that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-15 20:10 +0000 [r345220-345432] Richard Mudgett * /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. * Change from using send() to ast_agi_send() so the HANGUP shows up in the AGI debug output. (closes issue ASTERISK-18723) Reported by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 345431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure name. It is fortunate that the typo does not alter generated code since the e->restart.channel and e->ring.channel members are in the same position. (closes issue ASTERISK-18868) Reported by: zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by zvision ........ Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused for AMI and realtime. * Add parameter to queue log ADDMEMBER to indicate if the member is paused. (closes issue ASTERISK-18645) Reported by: garlew Patches: paused.diff (License #5337) patch uploaded by garlew Tested by: rmudgett, garlew Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt, channels/sip/include/sip.h: Restore SIP DTMF overlap dialing method. The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes issue ASTERISK-18857) Reported by: David M Patches: mainpbx-trivial.patch (License #6326) patch uploaded by David M ........ Merged revisions 345219 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 19:10 +0000 [r345164] Terry Wilson * main/channel.c, /: Don't read past end of input when calling write() int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when new_frames == 1. Otherwise we start reading into adjacent variables declared on the stack. The read end discards what is read, so the values don't matter but it's not a good idea to read past where we want even though new_frames is almost always 1 and should never be large. This patch is basically taken out of kpfleming's eventfd branch, as he mentioned that he remembered fixing it there when I talked to him about this issue. Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged revisions 345163 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 19:01 +0000 [r345161] Walter Doekes * channels/sip/include/reqresp_parser.h, /: Update reqresp_parser parse_uri doxygen comments. The issue mentioned in the bug report had been fixed recently by twilson. The reporter included this documentation fix. (closes issue ASTERISK-18572) Reported by: Richard Miller Patch by: Richard Miller (modified) ........ Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 16:12 +0000 [r345117] Jonathan Rose * apps/app_voicemail.c, /: Moves voicemail setup password entry to the end of the setup process. This change was made because forcegreeting and forcename settings in voicemail could be circumvented by hanging up after entering a password, because the only way voicemail currently observes whether a mailbox is new or not is by checking to see if the password is the same as the mailbox number or not. (closes issue ASTERISK-18282) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ ........ Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 15:10 +0000 [r345064] Kinsey Moore * /, channels/chan_sip.c: Ensure that a null vmexten does not cause a segfault When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault. ........ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-12 16:17 +0000 [r344966] Gregory Nietsky * channels/chan_misdn.c, /: mISDN Round Robin break when no channel is available Prevent channels been parsed repetitively. ........ Merged revisions 344965 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-12 00:34 +0000 [r344900] Terry Wilson * /, res/res_musiconhold.c: Don't forget to rescan MOH files for cached realtime classes Realtime MOH class caching was implemented because without it, you would build a completely new MOH class and would start the music over at the beginning each time hold was pressed in a conversation. Unfortunately, this broke re-scanning for file changes for realtime MOH classes. This patch corrects that issue. (closes issue ASTERISK-18039) Review: https://reviewboard.asterisk.org/r/1579/ ........ Merged revisions 344899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 21:58 +0000 [r344845] Walter Doekes * include/asterisk/stringfields.h, include/asterisk/utils.h, /, main/utils.c: Use __alignof__ instead of sizeof for stringfield length storage. Kevin P Fleming suggested that r343157 should use __alignof__ instead of sizeof. For most systems this won't be an issue, but better fix it now while it's still fresh. Review: https://reviewboard.asterisk.org/r/1573 ........ Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 21:50 +0000 [r344842] Matthew Jordan * /, main/file.c: Video format was treated as audio when removed from the file playback scheduler This patch fixes the format type check in ast_closestream and filestream_destructor. Previously a comparison operator was used, but since audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats that have a value greater than the video formats), a bitwise AND operation is used instead. Duplicated code was also moved to filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo Bedrij Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1580/ ........ Merged revisions 344823 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 21:36 +0000 [r344836-344839] Walter Doekes * /, channels/sip/reqresp_parser.c: Remove unneeded if(params) checks in reqresp_parser. Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent reason. There is no way that params could become NULL in that piece of code, so I removed these excess checks again. ........ Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /: Fix bad quoting of multiline mxml opaque_data that caused invalid xml. The opaque_data was added and enclosed in single quotes, assuming it would be only a single line. The rest of the lines were appended after the closing quote. (closes issue ASTERISK-18852) Reported by: peep_ on IRC Review: https://reviewboard.asterisk.org/r/1577 ........ Merged revisions 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 20:14 +0000 [r344770] Kinsey Moore * /, channels/chan_sip.c: Fix regression introduced by SDP fixups If capability is adjusted when switching to UDPTL during fax transmission, fax teardown fails. Make sure capability is only touched if RTP is active. This regression was introduced in R344385. ........ Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 18:36 +0000 [r344662-344716] Richard Mudgett * /, channels/chan_sip.c: Check sip.conf maxforwards parameter for range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/cli.c, /: Make CLI "core show channel" not hold the channel lock during console output. Holding the channel lock while the CLI "core show channel" command is executing can slow down the system. It could block the system if the console output is halted or paused. * Made capture the CLI "core show channel" output into a buffer to be output after the channel is unlocked. * Removed use of C++ keyword as a variable name. out renamed to obuf. * Checked allocation of obuf for failure so will not crash. (closes issue ASTERISK-18571) Reported by: Pavel Troller Tested by: rmudgett ........ Merged revisions 344661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 15:33 +0000 [r344609] Jonathan Rose * main/pbx.c, /: Fix a segmentation fault when using an extension with CID matching and no CID. Attempting to call an extension which used Caller ID matching with a channel that has an empty caller id string would result in a segmentation fault. (closes issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged revisions 344608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-10 23:21 +0000 [r344537-344557] Richard Mudgett * apps/app_macro.c: Fix app_macro.c MODULEINFO section termination. (closes issue ASTERISK-18848) Reported by: Tony Mountifield * /, apps/app_queue.c: Fix potential deadlock calling ast_call() with channel locks held. Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks held. Chan_local attempts to do deadlock avoidance in its ast_call() callback and could deadlock if a channel lock is already held. ........ Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel. It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-10 21:54 +0000 [r344493] David Vossel * main/bridging.c: Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision 2011-11-10 21:14 +0000 [r344386-344440] Kinsey Moore * /, apps/app_meetme.c: Fix another incorrect case with meetme's PIN logic and add documentation This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several bugs with SDP parsing and well-formedness of responses Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-10 16:25 +0000 [r344334] Matthew Nicholson * res/res_rtp_asterisk.c, /: only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified) ASTERISK-18490 ........ Merged revisions 344330 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 20:53 +0000 [r344271] Richard Mudgett * /, channels/chan_sip.c: Fix deadlock during dialplan reload. Another deadlock between the conlock/hints and channels/channel locking orders. * Don't hold the channel and private lock in sip_new() when calling ast_exists_extension(). (closes issue ASTERISK-18740) Reported by: Byron Clark Patches: sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark Tested by: Byron Clark ........ Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 20:07 +0000 [r344175-344216] Terry Wilson * channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c, channels/sip/reqresp_parser.c, channels/sip/include/sip.h: Don't treat a host:port string as a domain The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, tests/test_netsock2.c: Add a unit test for ast_sockaddr_split_hostport Review: https://reviewboard.asterisk.org/r/1575/ ........ Merged revisions 344157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 19:02 +0000 [r344159-344160] Alexandr Anikin * /: delete svn:mergeinfo * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c: Generate response to Status Enquiry message with Status q.931 message. Some PBXes require this for call status checking (closes issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches: ASTERISK-18748-5.patch (License #5415) patch uploaded by may213 Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 17:14 +0000 [r344103] Kinsey Moore * /, apps/app_meetme.c: Fix pin parameter behavior regression in MeetMe The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 15:26 +0000 [r344049] Matthew Nicholson * /, formats/format_wav.c: don't call ltohl() twice on the same value ASTERISK-18739 Patch by: pawel (modified) ........ Merged revisions 344048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-08 21:59 +0000 [r344004] Richard Mudgett * channels/chan_sip.c: Residual changes for Asterisk v10 branch from ASTERISK-18747. Residual changes for Asterisk v10 branch from ASTERISK-18747 after https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs callid hash key change fix. * Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. This is an optimization to avoid an unneeded lock/unlock and object search when using ao2_unlink. * Prevent crash in check_rtp_timeout() if dialog->rtp is NULL. Review: https://reviewboard.asterisk.org/r/1557/ 2011-12-15 Asterisk Development Team * Asterisk 10.0.0 Released. 2011-12-08 Asterisk Development Team * Asterisk 10.0.0-rc3 Released. * Add ASTSBINDIR to the list of configurable paths This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ * Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) * Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1610/ * Fix a change in behavior in 'database show' from 1.8. In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. (closes issue ASTERISK-18886) * Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ * Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ 2011-11-15 Asterisk Development Team * Asterisk 10.0.0-rc2 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK10RCS-3 * Ensure that a null vmexten does not cause a segfault Ensure that a null vmexten does not cause a segfault. When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision * Fix app_macro.c MODULEINFO section termination. (closes issue ASTERISK-18848) Reported by: Tony Mountifield 2011-11-08 Asterisk Development Team * Asterisk 10.0.0-rc1 Released. * Test results: http://bamboo.asterisk.org/browse/AST10-LUCID-317 2011-11-08 19:27 +0000 [r343944] wdoekes : * /, pbx/pbx_config.c: Fix crash when dialplan remove include is called with too few arguments. "dialplan remove include x from y" crashed when the amount of arguments was less than 6. (closes issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by: Andrey Solovyev ........ Merged revisions 343936 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-08 18:29 +0000 [r343900] David Vossel * channels/chan_sip.c: Fixes regression caused by r343635 There was a missing unlock for a function return that is only present in Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: Michael L. Young Patches: asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young 2011-11-08 18:01 +0000 [r343852] Richard Mudgett * /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect variable if unknown host configured crash. * Fixed a LOG_ERROR message referencing the config variable list v that had previously been processed and became NULL. * Added error return value set that was missing in an ast_append_ha() error return path. (closes issue ASTERISK-18743) Reported by: Michele Patches: issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes Tested by: Michele ........ Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-08 13:26 +0000 [r343789-343792] Leif Madsen * /: Recorded merge of revisions 343791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. * build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. 2011-11-07 22:37 +0000 [r343743] Kinsey Moore * channels/chan_sip.c: Make "sip show settings" CLI command get RPID flags from the right global page The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command pulled the flags from the incorrect global flags page. These are now read from sip global flags page 0. (closes issue AST-711) 2011-11-07 21:42 +0000 [r343691] Matthew Nicholson * /, channels/chan_sip.c: respect case changes in peer names on sip reload ASTERISK-18669 ........ Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 21:27 +0000 [r343677] Richard Mudgett * /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid. Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. ........ Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 20:31 +0000 [r343635] Kinsey Moore * /, channels/chan_sip.c: Prevent BLF subscriptions from causing deadlocks Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ ........ Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 19:55 +0000 [r343580] wdoekes : * main/udptl.c, UPGRADE.txt: Correct the default udptl port range. The udptl port range was defined as 4000-4999 in the udptl.conf.sample, as 4500-4599 if you didn't have a config and 4500-4999 if your config was broken. Default is now 4000-4999. (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher Review: https://reviewboard.asterisk.org/r/1565 2011-11-07 19:51 +0000 [r343578] Richard Mudgett * /, channels/chan_sip.c: Fix deadlock if peer is destroyed while sending MWI notice. A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. NOTE: This particular deadlock will not happen with Asterisk 10, but some of the changes still apply. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ ........ Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 18:39 +0000 [r343533] Matthew Nicholson * main/format.c: list all of the codecs associated with a particular format id for CLI command "core show codec" AST-699 2011-11-04 15:11 +0000 [r343445] Alexandr Anikin * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h, addons/ooh323c/src/printHandler.c: Final fix memleaks in GkClient codes, same for Timer codes. (these memleaks stop development of gk codes, now i can continue) Fix printHandler 'Unbalanced Structure' issues with locking printHandler data for single thread. ........ Merged revisions 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-03 20:31 +0000 [r343393] wdoekes : * /, res/res_config_sqlite.c: Fix sqlite config driver segfault and broken queries The sqlite realtime handler assumed you had a static config configured as well. The realtime multientry handler assumed that you weren't using dynamic realtime. (closes issue ASTERISK-18354) (closes issue ASTERISK-18355) Review: https://reviewboard.asterisk.org/r/1561 ........ Merged revisions 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-03 19:57 +0000 [r343337] Richard Mudgett * /, funcs/func_dialgroup.c: Remove invalid flag given to iterator in func_dialgroup.c ........ Merged revisions 343336 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-03 15:39 +0000 [r343221-343277] Terry Wilson * /, channels/sip/include/sip.h: Make room for the fax detect flags The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c, channels/sip/include/sip.h: Remove registertrying option in chan_sip This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-02 22:24 +0000 [r343158-343192] wdoekes : * /, channels/chan_sip.c: Fix improper warning introduced by r342927 and more tweaks Changeset r342927 introduced a warning which was only supposed to be emitted when a found realtime peer had an empty (or no) name. It turned out that there were some inconsistencies left. Now found peers with an empty name are explicitly ignored like before r342927 but better. Reviewed by: Stefan Schmidts, Terry Wilson Review: https://reviewboard.asterisk.org/r/1560 ........ Merged revisions 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * include/asterisk/stringfields.h, include/asterisk/utils.h, /, main/utils.c: Ensure that string field lengths are properly aligned Integers should always be aligned. For some platforms (ARM, SPARC) this is more important than for others. This changeset ensures that the string field string lengths are aligned on *all* platforms, not just on the SPARC for which there was a workaround. It also fixes that the length integer can be resized to 32 bits without problems if needed. (closes issue ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: https://reviewboard.asterisk.org/r/1549 ........ Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-02 19:33 +0000 [r343048-343103] Leif Madsen * /, apps/app_authenticate.c: Add note about how Authenticate() application with option 'd' works. (closes issue ASTERISK-17422) Reported by: Leif Madsen ........ Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configs/queues.conf.sample: Update documentation for leastrecent strategy. In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated to reflect how the strategy actually checks peers. (closes issue ASTERISK-17854) Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch (License #6139) ........ Merged revisions 343047 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-02 13:45 +0000 [r342991] Kevin P. Fleming * /, apps/app_meetme.c: Modify comments in MeetMe application documentation about DAHDI. The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-01 20:58 +0000 [r342870-342929] wdoekes : * /, channels/chan_sip.c, configs/extconfig.conf.sample, include/asterisk/config.h, main/config.c: Several fixes to the chan_sip dynamic realtime peer/user lookup There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to *still* have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/realtime/mysql/sipfriends.sql (removed), contrib/realtime/mysql/sippeers.sql (added), configs/res_config_mysql.conf.sample, /, configs/extconfig.conf.sample, configs/res_ldap.conf.sample, res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample, main/config.c: Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-31 17:46 +0000 [r342824] Richard Mudgett * main/format.c, main/format_cap.c: Misc format capability fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable. * Fix potential race between checking if an interface exists and adding it to the container in format.c:ast_format_attr_reg_interface(). * Fixed double rwlock destroy in format.c:ast_format_attr_init() error exit path. * Simplified format.c:find_interface() and format.c:has_interface(). 2011-10-31 16:04 +0000 [r342770] Matthew Jordan * main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access when adding extension to pattern match tree When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 ........ Merged revisions 342769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-30 02:21 +0000 [r342715] Terry Wilson * res/res_calendar.c: Don't crash on empty notify channel 2011-10-29 04:26 +0000 [r342662] Richard Mudgett * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-27 20:10 +0000 [r342605] Matthew Nicholson * main/dsp.c: tweak the v21 detector to detect an additional pattern of hits and misses 2011-10-27 19:41 +0000 [r342546-342603] Jonathan Rose * res/res_rtp_multicast.c, /: Fix sequence number overflow over 16 bits causing codec change in RTP packets. Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_jabber.c: Cleanup reference leaks in res_jabber res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-25 22:05 +0000 [r342485-342488] Richard Mudgett * /, main/astobj2.c: Check fopen return value for ao2 reference debug output. Reported by: wdoekes Patched by: wdoekes Review: https://reviewboard.asterisk.org/r/1539/ ........ Merged revisions 342487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_pri.c: Change D-channel warning to be less confusing on non-NFAS setups. The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-25 21:10 +0000 [r342381-342436] Terry Wilson * /, apps/app_queue.c: Use int for storing ao2_container_count instad of size_t AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Simplify queue membercount code Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Properly update membercount for reloaded members Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-25 19:08 +0000 [r342277-342329] Kinsey Moore * pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for pbx_spool.c One of the changes in the recent spool handling of hardlinks patch was just outside a HAVE_INOTIFY block and caused compilation to fail in some build environments. This has been corrected. ........ Merged revisions 342328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * pbx/pbx_spool.c, /: Merged revisions 342276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | 18 lines Fix spool handling to allow call files to be hardlinked into place This fixes the inotify code to handle call files being hardlinked into the spool directory. The smsq utility does this, instead of rename(), to ensure that it cannot accidentally overwrite an existing spool file. A rename() might do that, but link() will definitely not. The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE event on the file... which was never forthcoming, since it was never opened. Now we look for IN_OPEN events following the IN_CREATE event, and only wait for an IN_CLOSE_WRITE if the file was actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: https://reviewboard.asterisk.org/r/1391/ ........ 2011-10-25 01:25 +0000 [r342224] Terry Wilson * /, include/asterisk/config.h, main/config.c: Return NULL when no results returned for realtime_multientry It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-24 22:32 +0000 [r342183] Richard Mudgett * include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines. 2011-10-24 19:51 +0000 [r342062] Jonathan Rose * /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now include fromuser of related peer. This behavior matches up more closely with the way invite/register/etc are handled. This patch also modifies some adjacent code for code style compliance. Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232) ........ Merged revisions 342061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-24 07:31 +0000 [r341920-342017] Gregory Nietsky * apps/app_queue.c: queues container needs locking when using the OBJ_NOLOCK flag * apps/app_queue.c: Remove some ref leaks and a return without unlock. There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. * /, apps/app_queue.c: Revert Janitor patch 341920 For now * /, apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial patch is related to work on RB1538 ........ Merged revisions 341906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-21 16:42 +0000 [r341807-341810] Matthew Nicholson * /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395 ........ Merged revisions 341809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, pbx/pbx_lua.c: don't limit the length of app and function arguments ASTERISK-18395 ........ Merged revisions 341806 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-20 21:58 +0000 [r341718] Richard Mudgett * include/asterisk/features.h, /, main/features.c, res/res_agi.c: Fix AGI exec Park to honor the Park application parameters. The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-20 21:27 +0000 [r341665-341707] Paul Belanger * /, funcs/func_callerid.c: Fixed typo from previous commit ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_callerid.c: Updated documentation for the optional CID parameter with CALLERID ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-20 18:20 +0000 [r341580-341599] Gregory Nietsky * configs/queues.conf.sample: add documentation for check_state_unknown in configs/queues.conf.sample app_queue allows calls to members in a "Unknown" state to be treated as available setting check_state_unknown = yes will cause app_queue to query the channel driver to better determine the state this only applies to queues with ringinuse or ignorebusy set appropriately. * CHANGES, apps/app_queue.c: Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ 2011-10-20 15:14 +0000 [r341530] Terry Wilson * /, include/asterisk/strings.h: Clean up ast_check_digits The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-19 21:23 +0000 [r341486] Matthew Nicholson * apps/app_queue.c: Fix a performance regression introduced in r325483. The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 2011-10-19 19:01 +0000 [r341436] Paul Belanger * /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-19 07:42 +0000 [r341380] Terry Wilson * /, channels/chan_sip.c, include/asterisk/strings.h: Don't use is_int() since it doesn't link well on all platforms Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-19 07:23 +0000 [r341377] Stefan Schmidt * /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-18 23:42 +0000 [r341315] Terry Wilson * /, channels/chan_sip.c: Don't resolve numeric hosts or contact unresolved hosts If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-18 23:33 +0000 [r341313] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 341312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines fix issue on channel numbering (calls could have same channel number on heavy loaded system) ........ 2011-10-18 21:11 +0000 [r341255] Richard Mudgett * channels/chan_mgcp.c, include/asterisk/features.h, channels/chan_dahdi.c, channels/sig_analog.c, /, channels/chan_sip.c, main/features.c, channels/chan_iax2.c, channels/sip/include/sip.h: More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-17 17:36 +0000 [r341190] Terry Wilson * /, channels/chan_sip.c: Initialize variables before calling parse_uri If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-17 16:53 +0000 [r341148] Tzafrir Cohen * /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. Merged-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 2011-10-17 16:38 +0000 [r341122-341146] Paul Belanger * tests/test_format_api.c: Set 'core' support level for test_format_api.c * apps/app_voicemail.c, /: Multiple revisions 341108,341112 ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-17 16:18 +0000 [r341094] Jason Parker * CHANGES: Add information about limitations of new codec support in channel drivers. (issue ASTERISK-18680) 2011-10-17 15:39 +0000 [r341089] Terry Wilson * /, channels/chan_sip.c: Don't try to remove peers without IPs from peers_by_ip (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-14 21:36 +0000 [r341023] Kevin P. Fleming * /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change the internal name of the menuselect options that are used to control whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-14 20:50 +0000 [r340971] Kinsey Moore * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ 2011-10-14 18:23 +0000 [r340931] Jonathan Rose * utils/utils.xml, funcs/func_jitterbuffer.c: Some additional module documentation changes for 10 for the menuselect change. (issue ASTERISK-18268) 2011-10-14 16:39 +0000 [r340879] Terry Wilson * main/channel.c, /: Avoid unnecessary WARNING message Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-14 16:18 +0000 [r340868] Jonathan Rose * funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml, /, res/res_fax.c, apps/app_celgenuserevent.c, codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c: Fixes some support level info so that it can be read by menuselect. (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ ........ Merged revisions 340863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-13 22:54 +0000 [r340810] Richard Mudgett * /, main/features.c: Fix DTMF blind transfer continuing to execute dialplan after transfer. Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-13 08:46 +0000 [r340770] Gregory Nietsky * channels/chan_sip.c: Only send MWI Notify on register if the registration is successful. lastmsgssent was removed from chan_sip and the old behavior of sending a mwi notify on register [except when subscribemwi is set] was restored but this must only happen when registration succeeds. leaking information for unsuccessful registrations is not secure. 2011-10-13 06:59 +0000 [r340718] Stefan Schmidt * channels/chan_sip.c: Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ 2011-10-13 06:56 +0000 [r340578-340716] Terry Wilson * /, channels/chan_sip.c: Initialize ast_sockaddr before calling ast_sockaddr_resolve Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_config_sqlite.c: Don't skip the query field on a realtime multi query There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ 2011-10-12 20:33 +0000 [r340577] Stefan Schmidt * /, channels/chan_sip.c: Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ 2011-10-12 20:08 +0000 [r340471-340523] Richard Mudgett * channels/chan_dahdi.c, /: Initialize the PRI channel alarms properly on startup. The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_meetme.c: Update MeetMe p and X option documentation when interacting with the s option. ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-12 16:28 +0000 [r340419] Paul Belanger * /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was added (closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-11 21:05 +0000 [r340281-340366] Richard Mudgett * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: Add protection for SS7 channel allocation and better glare handling. * Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some potential deadlocks pointed out by helgrind. * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /, include/asterisk/manager.h: Convert registered AMI actions to ao2 objects. * Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-11 18:41 +0000 [r340280] Tzafrir Cohen * main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 2011-10-10 22:55 +0000 [r340219-340222] Terry Wilson * main/db.c: On astdb conversion, also warn about permissions requirements The user running Asterisk must have permission to the directory the Asterisk database resides in since SQLite 3 needs to be able to create a journal file. (closes issue ASTERISK-18174) * utils/astdb2bdb.c (added): Add a missing file for the astdb2bdb conversion utility * utils/Makefile, utils/utils.xml, UPGRADE.txt: Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ 2011-10-10 20:30 +0000 [r340165] Matthew Jordan * /, channels/chan_sip.c: Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ 2011-10-10 14:15 +0000 [r340109] Matthew Nicholson * main/loader.c, main/xmldoc.c, main/pbx.c, main/manager.c, /, res/res_fax.c, apps/app_fax.c, include/asterisk/module.h, res/res_agi.c, include/asterisk/xmldoc.h, doc/appdocsxml.dtd: Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ 2011-10-09 22:18 +0000 [r339992-340031] Damien Wedhorn * channels/chan_skinny.c: Return -1 to skinny_session if register rejected. If device registration is rejected, return -1 so that the session is destroyed immediately. Previously, a segfault would occur on a graceful shutdown if a register is rejected and the skinny_session has not yet timed out. * channels/chan_skinny.c: Remove log message on traverse session list. On destroying a session, a list of sessions is traversed to find the matching session. For each session not matching, skinny erroneously logged that the session was not matched. While technically correct the message was misleading, and tended to indicate errors that were not there. 2011-10-09 01:18 +0000 [r339831-339942] igorg : * channels/chan_unistim.c, /: Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ * channels/chan_unistim.c, /: Merged revisions 339884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ * channels/chan_unistim.c, /: Merged revisions 339830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ 2011-10-07 19:36 +0000 [r339777] Richard Mudgett * /, apps/app_url.c: Merged revisions 339776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia ........ 2011-10-06 23:08 +0000 [r339722] Damien Wedhorn * channels/chan_skinny.c: Reject v17 skinny devices in Asterisk10 Small fix for Asterisk10 to reject skinny devices with skinny firmware version17 and above. Review: https://reviewboard.asterisk.org/r/1497/ 2011-10-06 22:58 +0000 [r339720] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, configure.ac, autoconf/ast_ext_lib.m4: Merged revisions 339719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri *must* have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ 2011-10-06 20:47 +0000 [r339681] Damien Wedhorn * channels/chan_skinny.c: Fixed segfault on core stop gracefully. There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff (in trunk) and myself. 2011-10-06 17:53 +0000 [r339626] Richard Mudgett * main/udptl.c, /, channels/chan_sip.c: Merged revisions 339625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ 2011-10-06 13:43 +0000 [r339586] Leif Madsen * /, build_tools/prep_tarball: Merged revisions 339566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) | 8 lines Update prep_tarball script to download pre-exported documentation. I've updated the prep_tarball script to now download the pre-exported documentation from the Asterisk wiki. This will give us more control over what is being included in the tarball releases, and will make both the PDF and HTML exported documentation look much better (especially when viewing from a console). (Closes issue ASTERISK-18677) ........ 2011-10-05 17:01 +0000 [r339508-339512] Richard Mudgett * apps/app_dial.c, /: Merged revisions 339511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line Fix Dial F option notes formatting. ........ * main/manager.c, /: Merged revisions 339504,339506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) | 7 lines Add missing documentation of required AMI action Challenge AuthType header. (closes issue ASTERISK-18554) Reported by: Vlad Povorozniuc Patches: __20110919-manager-challenge-docs.patch.txt (license #4999) patch uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI action Challenge. ........ 2011-10-05 16:32 +0000 [r339507] Matthew Nicholson * /, res/res_fax.c: Merged revisions 339505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines The app name in the documentation must match what we register the application as. ........ 2011-10-05 06:28 +0000 [r339463] Gregory Nietsky * res/res_fax.c: Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. 2011-10-04 22:56 +0000 [r339407] Richard Mudgett * Makefile, /: Merged revisions 339406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) | 8 lines Make always create the MOH directory (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael Keuter ........ 2011-10-04 19:44 +0000 [r339298-339353] Jonathan Rose * /, main/say.c: Merged revisions 339352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | 12 lines Removes improper use of sound 'and' in German language mode from application saynumber Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk will only say the 'and' word between the single digit and double digit places. (closes issue ASTERISK-18212) Reported By: Lionel Elie Mamane Patches: upstream_germand_no_and.diff (License #5402) uploaded by Lionel Elie Mamane ........ * /, res/res_jabber.c: Merged revisions 339297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines Reverting revision 333265 due to component connection problems it introduces. I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this problem, but first it seems prudent to remove this rather broad attempt to fix it and instead approach this problem either from the same angle but looking only at canceling (or possibly rescheduling) the send when we absolutely know it will cause a segfault or, if that can't be easily accomplished, strictly from the devstate side of things. Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe. (issue ASTERISK-18626) (issue ASTERISK-18078) ........ 2011-10-04 11:49 +0000 [r339245] Alexandr Anikin * addons/ooh323c/src/memheap.c, /: Merged revisions 339244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 lines fix forget declaration in previous change ........ 2011-10-03 20:13 +0000 [r339145-339148] Leif Madsen * /, channels/chan_sip.c: Merged revisions 339147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) | 6 lines Remove duplicated Maxforwards line in AMI output. (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny ........ * apps/app_dial.c, /: Merged revisions 339144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines Make documentation for Dial() options 'F' and 'F()' more clear. (Closes issue ASTERISK-18646) Reported by: Physis Heckman Tested by: Richard Mudgett ........ 2011-10-03 18:52 +0000 [r339089] Alexandr Anikin * addons/ooh323c/src/memheap.c, /: Merged revisions 339087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 lines destroy memheap mutex properly before memheap deleted (fix memory leak occured after r304950 changes with DEBUG_THREAD compile option) ........ 2011-10-03 18:44 +0000 [r339088] Terry Wilson * /, channels/chan_sip.c, main/file.c: Merged revisions 339086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ 2011-10-03 15:54 +0000 [r339011-339045] Matthew Nicholson * res/res_fax.c: Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here. This function prints a list of caps instead of a hex bitfield. * res/res_fax.c: Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. * res/res_fax.c: properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags) 2011-10-03 14:38 +0000 [r338904-338997] Gregory Nietsky * CHANGES: Documentation noting the extension of CHANNEL() for chan_ooh323 * addons/chan_ooh323.c, funcs/func_channel.c: Remove the channel function OOH323() and place its options into CHANNEL() channel drivers should not have there own dialplan functions. * res/res_fax.c: Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will turn off the gateway but the framehook is not destroyed. this problem happens when a gateway is attempted in the dialplan and the device is not available i may want to do fax to mail in the server it will not be allowed. instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts 338904 Fix some white space. * res/res_fax.c: Remove T38 Gateway capability when detaching framehook. SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. 2011-09-30 22:06 +0000 [r338801] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 338800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines Fix segfault in analog_ss_thread() not checking ast_read() for NULL. NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch ........ 2011-09-30 18:55 +0000 [r338719] Jonathan Rose * /, configs/queues.conf.sample: Merged revisions 338718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) | 1 line Adds documentation for QueueMemberStatus event generation ........ 2011-09-30 16:35 +0000 [r338664] Richard Mudgett * /, channels/chan_sip.c: Fix formatting of AMI header for SIP show peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-09-29 21:14 +0000 [r338556] Paul Belanger * tests/test_amihooks.c, tests/test_security_events.c, /, tests/test_locale.c, tests/test_logger.c, tests/test_dlinklists.c, tests/test_linkedlists.c: Merged revisions 338555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) | 2 lines Test modules should depend on the TEST_FRAMEWORK flag ........ 2011-09-29 20:54 +0000 [r338552] Jason Parker * /, tests/test_db.c, tests/test_netsock2.c: Merged revisions 338551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line Test modules have a support level of core. ........ 2011-09-29 18:32 +0000 [r338493] Leif Madsen * /, channels/chan_sip.c: Merged revisions 338492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines Update documentation for SIP_HEADER. The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) ........ 2011-09-29 12:16 +0000 [r338417] Gregory Nietsky * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ 2011-09-28 22:36 +0000 [r338253-338323] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 338322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ * include/asterisk/logger.h, /: Merged revisions 338235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ 2011-09-28 20:54 +0000 [r338228] Jason Parker * build_tools/cflags.xml, channels/chan_usbradio.c, build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /, build_tools/embed_modules.xml, tests/test_db.c, tests/test_netsock2.c: Merged revisions 338227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ 2011-09-28 20:26 +0000 [r338225] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 338224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ 2011-09-27 20:13 +0000 [r338085] Paul Belanger * /, apps/app_macro.c: Merged revisions 338084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) | 2 lines Upgrade app_macro to core ........ 2011-09-26 19:35 +0000 [r337974] Richard Mudgett * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /, include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c, include/asterisk/channel.h, main/cel.c, main/manager.c, funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c, main/logger.c, cel/cel_sqlite3_custom.c: Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ 2011-09-27 Asterisk Development Team * Asterisk 10.0.0-beta2 Released. * Based on revision that passed automated testing (http://bamboo.asterisk.org/browse/AST10-LUCID-178) 2011-09-26 19:35 +0000 [r337974] Richard Mudgett * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /, include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c, include/asterisk/channel.h, main/cel.c, main/manager.c, funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c, main/logger.c, cel/cel_sqlite3_custom.c: Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ 2011-09-23 19:18 +0000 [r337840-337902] Gregory Nietsky * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | 4 lines Spelling fix ........ * /, apps/app_queue.c: Merged revisions 337839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 ........ 2011-09-23 00:45 +0000 [r337775] Russell Bryant * configs/res_pktccops.conf.sample, /: Merged revisions 337774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) | 11 lines Comment out entries in sample res_pktccops.conf. With these options enabled, they can cause Asterisk to freak out by SYN flooding a network and eating the CPU. Obviously it would be good to fix the code so that this can't happen, but we can at least change the default configuration so it doesn't happen. This was reported downstream to the Fedora issue tracker: https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........ 2011-09-22 21:37 +0000 [r337721] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ 2011-09-22 18:43 +0000 [r337640] Paul Belanger * CREDITS, apps/app_meetme.c, CHANGES: Revert previous commit New feature should be added into trunk, unfortunately it is too late for the Asterisk 10 branch. 2011-09-22 15:47 +0000 [r337595-337597] Jonathan Rose * channels/sip/security_events.c (added), channels/sip/include/security_events.h (added): Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ * configs/logger.conf.sample, channels/chan_sip.c, include/asterisk/event_defs.h, main/security_events.c, main/event.c, CHANGES, channels/sip/include/sip.h, include/asterisk/security_events_defs.h: Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ 2011-09-22 11:44 +0000 [r337431-337542] Gregory Nietsky * res/res_srtp.c, /: Merged revisions 337541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ * /, channels/chan_h323.c: Merged revisions 337486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ * apps/app_originate.c, CHANGES: Revert commit r337261 This commit is for trunk not version 10 ----- Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. ----- * main/channel.c, /: Merged revisions 337430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines Its possible to loose audio on ast_write when the channel is not transcoded correctly. in the case of DAHDI the channel is hungup. This patch tries to "fix" the problem and make the channel compatiable and warn the user of this problem. Please note there is a underlying problem with codec negotion this does not fix the problem it does try to rectify it and prevent loss of service. Review: https://reviewboard.asterisk.org/r/1442/ (closes issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) (issue ASTERISK-18422) ........ 2011-09-21 21:25 +0000 [r337342-337380] Tilghman Lesher * apps/app_voicemail.c, /: More silly spacing changes ..... Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: ........ Dumb little spacing fix. ........ Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * funcs/func_curl.c, /: ........ Escape commas in keys and values, when keys and values are enumerated by commas. Review: https://reviewboard.asterisk.org/r/1433 ........ Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8 2011-09-21 11:15 +0000 [r337261-337263] Gregory Nietsky * configs/sip.conf.sample: Whitespace fixup from SRTP patch * apps/app_originate.c, CHANGES: Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ 2011-09-21 09:32 +0000 [r337178-337219] Olle Johansson * configs/extensions.conf.sample, main/pbx.c, CHANGES: Make ast_pbx_run() not default to s@default if extension is not found Review: https://reviewboard.asterisk.org/r/1446/ This is a bug - or architecture mistake - that has been in Asterisk for a very long time. It was exposed by the AMI originate action and possibly some other applications. Most channel drivers checks if an extension exists BEFORE starting a pbx on an inbound call, so most calls will not depend on this issue. Thanks everyone involved in the review and on IRC and the mailing list for a quick review and all the feedback. (closes issue ASTERISK-18578) * res/res_rtp_asterisk.c, configs/rtp.conf.sample, CHANGES: Change strictrtp option to default to yes in the RTP module Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c 2011-09-20 22:49 +0000 [r337120] Matthew Jordan * apps/app_voicemail.c, apps/app_dial.c, include/asterisk/app.h, /, apps/app_meetme.c, apps/app_minivm.c, main/app.c, apps/app_confbridge.c, apps/app_followme.c: Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ 2011-09-20 22:47 +0000 [r337119] Richard Mudgett * funcs/func_strings.c: Fix crash with STRREPLACE function. The ast_func_read() function calls the .read2 callback with the len parameter set to zero indicating no size restrictions on the supplied ast_str buffer. The value was used to dimension a local starts[] array with the array subsequently used. * Reworked the strreplace() function to perform the string replacement in a straight forward manner. Eliminated the need for the starts[] array. (closes issue ASTERISK-18545) Reported by: Federico Alves Patches: jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Federico Alves 2011-09-20 22:19 +0000 [r337116] Leif Madsen * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines Update RedHat Init script to work with Heartbeat. The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that it can work correctly with Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa ........ 2011-09-20 21:05 +0000 [r337062] Kinsey Moore * tests/test_pbx.c, main/pbx.c, /: Merged revisions 337061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | 11 lines Make CANMATCH with the new pattern match engine behave more like the old one When checking an extension for E_CANMATCH using the new extension matching algorithm, an exact match was not returned as a possible match resulting in the queue failing to allow a caller to exit on DTMF. This removes the requirement that an extension be longer than acquired digits for an E_CANMATCH operation to succeed. (closes issue ASTERISK-18044) Review: https://reviewboard.asterisk.org/r/1367/ ........ 2011-09-20 19:12 +0000 [r336978-337008] Richard Mudgett * /, channels/sig_ss7.c: Merged revisions 337007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) ........ * /, channels/sig_ss7.c: Merged revisions 336977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines Fix deadlock from not releasing SS7 linkset lock. sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ 2011-09-20 16:51 +0000 [r336936] Gregory Nietsky * channels/sip/sdp_crypto.c, channels/chan_sip.c, channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ 2011-09-20 01:03 +0000 [r336878] Russell Bryant * res/res_rtp_asterisk.c, /: Merged revisions 336877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ ........ 2011-09-19 22:13 +0000 [r336792] Terry Wilson * /, channels/chan_sip.c: Merged revisions 336791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ 2011-09-19 21:41 +0000 [r336734-336789] Tilghman Lesher * funcs/func_strings.c: Ensure substring will not be found in the previous match. * include/asterisk/optional_api.h, Makefile, /, configure, include/asterisk/autoconfig.h.in, main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules: Merged revisions 336733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. ........ 2011-09-19 20:16 +0000 [r336717] Jonathan Rose * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c, apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c: Merged revisions 336716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ 2011-09-19 18:51 +0000 [r336659] Richard Mudgett * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 ........ 2011-09-19 15:42 +0000 [r336573] Leif Madsen * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines Update get_ilbc_source.sh script to work again. Recently iLBC support in Asterisk has changed after the acquisition of GIPS by Google. More information about how this may affect you is available in a blog post at: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ ........ 2011-09-19 15:32 +0000 [r336570] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 336569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ 2011-09-19 13:48 +0000 [r336502-336504] Olle Johansson * Makefile: Revert accidental change * Makefile, /, channels/chan_sip.c: Merged revisions 336501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ 2011-09-19 13:31 +0000 [r336500] Gregory Nietsky * /, channels/chan_h323.c: Merged revisions 336499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines A long time ago in a galaxy far far away a IPv6 update was made, chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ 2011-09-19 12:15 +0000 [r336381-336441] Olle Johansson * main/manager.c, /: Merged revisions 336440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines Make sure manager_debug option is reset at reload ........ * /, channels/chan_sip.c: Merged revisions 336378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ 2011-09-16 22:11 +0000 [r336313-336316] Terry Wilson * /, funcs/func_frame_trace.c: Merged revisions 336314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines Whitespace fix ........ * /, funcs/func_frame_trace.c: Merged revisions 336312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines Add missing frame types to func_frame_trace Also casts control frames to the proper enum so that the compile will catch new additions. ........ 2011-09-16 21:09 +0000 [r336307] Jonathan Rose * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/frame.h: Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ 2011-09-16 19:10 +0000 [r336235] Sean Bright * /, UPGRADE-1.8.txt: Merged revisions 336234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines Make a note that inotify won't work with an NFS mounted spooler directory. ........ 2011-09-16 10:12 +0000 [r336094-336167] Gregory Nietsky * channels/chan_misdn.c, /: Merged revisions 336166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines The round robin routing routine in chan_misdn.c is broken. it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ ........ * /, apps/app_queue.c: Merged revisions 336093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines Locking order in app_queue.c causes deadlocks. a channel lock must never be held with the queues container lock held. the deadlock occured on masquerade. the queues container lock is a relic of the past the old queue module lock. with ao2 there is no need to hold this lock when dealing with members this patch removes unneeded locks. (closes issue ASTERISK-18101) (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew Nicholson Review: https://reviewboard.asterisk.org/r/1402/ ........ 2011-09-15 15:19 +0000 [r336091] David Vossel * main/format_cap.c: Removes some no-op code found in format_cap.c. 2011-09-15 12:46 +0000 [r336042] Olle Johansson * CREDITS, apps/app_meetme.c, CHANGES: Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden 2011-09-15 08:29 +0000 [r335991] Gregory Nietsky * /, channels/chan_agent.c: Merged revisions 335978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines lock the channel before calling ast_bridged_channel() to prevent a seg fault. AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot ........ 2011-09-14 18:31 +0000 [r335852-335912] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 335911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines Remove unnecessary libpri dependency checks in the configure script. Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter ........ * channels/chan_dahdi.c, /: Merged revisions 335851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines Fixed cut-n-paste regression using the wrong variable. Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett ........ 2011-09-14 13:28 +0000 [r335791] Matthew Nicholson * main/manager.c, /: Merged revisions 335790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines The tech and data members of fast_originate_helper are not string fields. ASTERISK-17709 ........ 2011-09-13 22:10 +0000 [r335721] Richard Mudgett * /, apps/app_directed_pickup.c: Merged revisions 335720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line Remove obsolete todo comment about PICKUPRESULT. ........ 2011-09-13 21:37 +0000 [r335717] Tzafrir Cohen * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do parse the option "defaultlanguage" from the [options] section of asterisk.conf, as in the sample config file. Otherwise the build-time default language (normally "en") is always the default one. Review: https://reviewboard.asterisk.org/r/1342/ Signed-off-by: Tzafrir Cohen (License #5035) Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716 2011-09-13 18:55 +0000 [r335656] Tilghman Lesher * /, configure, configure.ac: Merged revisions 335655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines Move mandatory checks closer to the beginning of the file. If these are going to fail, they should fail as quickly as possible. ........ 2011-09-13 18:47 +0000 [r335653] Matthew Nicholson * main/pbx.c, main/manager.c, /: Merged revisions 335618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines Don't limit the size of appdata for manager originate actions. ASTERISK-17709 Patch by: tilghman (with modifications) ........ 2011-09-13 07:24 +0000 [r335510] Russell Bryant * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged revisions 335497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ 2011-09-12 15:55 +0000 [r335434] Matthew Nicholson * main/channel.c, /: Merged revisions 335433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines Properly set caller_warning and callee_warning before we try to use them. ASTERISK-18199 Patch by: elguero Testing by: rtang ........ 2011-09-12 14:22 +0000 [r335346] Kinsey Moore * apps/app_dial.c, /: Merged revisions 335341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines Ensure frames are not written to dialed channel if ringback is requested When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) ........ 2011-09-12 13:47 +0000 [r335323] Olle Johansson * /, channels/chan_sip.c: Merged revisions 335319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ 2011-09-12 13:27 +0000 [r335321] Kinsey Moore * /, channels/chan_iax2.c: Merged revisions 335320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) ........ 2011-09-12 11:11 +0000 [r335260] Stefan Schmidt * /, channels/chan_sip.c: Merged revisions 335259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ 2011-09-09 16:27 +0000 [r335078] Matthew Jordan * channels/chan_mgcp.c, channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, addons/chan_ooh323.c, /, channels/chan_sip.c, channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c, main/channel.c, channels/chan_usbradio.c, main/dial.c, channels/chan_dahdi.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_frame_trace.c, main/features.c, channels/chan_h323.c, channels/chan_alsa.c, include/asterisk/frame.h, channels/sig_ss7.c: Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ 2011-09-09 07:23 +0000 [r335014] Gregory Nietsky * funcs/func_dialplan.c, apps/app_readexten.c, CHANGES: Move code for VALID_EXTEN from app_readexten to func_dialplan Mark VALID_EXTEN deprecated. Review: https://reviewboard.asterisk.org/r/1396/ 2011-09-08 22:28 +0000 [r334954] Richard Mudgett * /, main/logger.c: Merged revisions 334953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to unregister its logger level. * Make ast_logger_unregister_level() use ast_free() instead of free(). When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call to free(). Therefore, if you allocated memory with a form of ast_malloc you must free it with ast_free. ........ 2011-09-07 19:37 +0000 [r334844] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 334843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines Cleanup chan_iax2.c log messages Review: https://code.asterisk.org/code/cru/CR-AST-11 ........ 2011-09-07 19:33 +0000 [r334841] Richard Mudgett * /, main/features.c: Merged revisions 334840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines Fix AMI action Park crash. * Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 ........ 2011-09-07 15:10 +0000 [r334682-334747] Stefan Schmidt * /, main/features.c: Merged revisions 334682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ * main/features.c: another clean up * main/features.c: Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. 2011-09-07 08:14 +0000 [r334617-334621] Alec L Davis * /, CHANGES, apps/app_queue.c: Merged revisions 334620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines peroid typo ........ * main/pbx.c, /: Merged revisions 334616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines Prevent segfault if call arrives before Asterisk is fully booted. Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk is fully booted. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1407/ ........ 2011-09-06 15:47 +0000 [r334514] Paul Belanger * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: authdebug is now disabled by default To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ 2011-09-06 13:58 +0000 [r334455] Gregory Nietsky * apps/app_voicemail.c, /: Merged revisions 334453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ 2011-09-02 21:08 +0000 [r334297-334357] Richard Mudgett * /, res/res_musiconhold.c: Merged revisions 334355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines MusicOnHold has extra unref which may lead to memory corruption and crash. The problem happens when a call is disconnected and you had started a MOH class that does not use the files mode. If you define REF_DEBUG and recreate the problem, it will announce itself with the following warning: Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, and class is still in a container! * Fixed moh_alloc() and moh_release() functions not handling the state->class reference consistently. (closes issue ASTERISK-18346) Reported by: Mark Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski Review: https://reviewboard.asterisk.org/r/1404/ ........ * /, include/asterisk/config.h, main/config.c: Merged revisions 334296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines Fix potential memory allocation failure crashes in config.c. * Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ ........ 2011-09-01 17:39 +0000 [r334235] Tilghman Lesher * main/pbx.c, /: Merged revisions 334234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011) | 2 lines Remove 1.6 compatibility documentation from 1.8, as it no longer applies. ........ 2011-09-01 17:36 +0000 [r334233] Matthew Nicholson * CHANGES: fixed a typo 2011-09-01 17:30 +0000 [r334230] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 334229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines Create a local alias for ast_odbc_clear_cache. As a function pointer, the reference has to be resolved at load time irrespective of the RTLD_LAZY flag. Creating a local alias solves this problem, because the structure is initialized with that local function pointer, while the actual function can remain lazily linked until runtime. The reason why this is important is because we lazily load function references during the module loading process, in order to obtain priority values for each module, ensuring that modules are loaded in the correct order. Previous to this change, when this module was initially loaded, the module loader would emit a symbol resolution error, because of the above requirement. Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by Walter Doekes, patch by me) ........ 2011-08-31 18:53 +0000 [r334064-334157] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 334156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ * res/res_fax.c: only alter the gateway_timeout when attching the gateway to a channel ASTERISK-18219 2011-08-31 16:00 +0000 [r334010-334013] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 334012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines No DAHDI channel available for conference, user introduction disabled. The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards Patches: jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ * main/channel.c, /, channels/chan_agent.c: Merged revisions 334009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines Call pickup race leaves orphaned channels or crashes. Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ ........ 2011-08-31 15:19 +0000 [r334007] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 334006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ 2011-08-30 21:53 +0000 [r333961-333962] Alexandr Anikin * addons/ooh323c/src/ooh323.c: security fix. really drop call if signalling addr is not same as socket addr * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c: Merged revisions 333947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines cleanups in ACF/ARJ GK replies processing fixed long (24 sec) pause if acf/arj proccessed before ast_cond_wait called to wait this ........ 2011-08-30 14:01 +0000 [r333895] Matthew Nicholson * res/res_fax.c: Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway). Patch by: irroot Review: https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 2011-08-29 21:41 +0000 [r333837] Terry Wilson * /, channels/chan_sip.c: Merged revisions 333836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ 2011-08-29 21:12 +0000 [r333786] Richard Mudgett * /, include/asterisk/channel.h, addons/chan_mobile.c: Merged revisions 333784-333785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines Fix deadlock potential of chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do not hold locks notes to channel.h ........ 2011-08-29 18:22 +0000 [r333716] Matthew Nicholson * res/res_fax_spandsp.c: It is possible for the gateway to be attached when the channel is still negotiating T.38. This change handles that case. ASTERISK-18329 2011-08-29 17:28 +0000 [r333681] Terry Wilson * main/channel.c, CHANGES: Use realtime text when it is negotiated This patch make use of wirte_text() realtime text instead of send_text() if T.140 is in native formats. ASTERISK-17937 Review: https://reviewboard.asterisk.org/r/1356/ 2011-08-29 17:12 +0000 [r333631] Matthew Jordan * apps/app_voicemail.c, /: Merged revisions 333630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI message in app_voicemail ........ 2011-08-29 15:56 +0000 [r333570] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | 4 lines Accidental use of variable client->status instead of client->state in from ASTERISK-18078 (issue ASTERISK-18078) ........ 2011-08-28 09:55 +0000 [r333508] Tzafrir Cohen * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6) GCC 4.6 detects variables that get assined to, but never used later. Also removes some remmed-out lines that become invalid. (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen (License #5035) , 2011-08-26 16:28 +0000 [r333410] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines [patch] Buddies are always auto-registered when processing the roster Reporter said autoregister flag was ignored for registering 'buddies' which had a subscription to us. Verified that this was the case and observed how the patch addressed this and made sure it didn't break anything. (closes issue ASTERISK-14233) Reported by: Simon Arlott Patches: asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott Tested by: Jonathan Rose ........ 2011-08-26 15:58 +0000 [r333370] Matthew Jordan * apps/app_voicemail.c, /: Merged revisions 333339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) | 20 lines Bug fixes for voicemail user emailsubject / emailbody. This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 ........ 2011-08-25 19:01 +0000 [r333268] Jason Parker * Makefile, /: Merged revisions 333267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) | 2 lines Fix for DESTDIR spaces patch. ........ 2011-08-25 19:00 +0000 [r333266] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines Segfault when publishing device states via XMPP and not connected When using publishing device state with res_jabber, Asterisk will attempt to send a device state using the unconnected client using iks_send_raw and crash. This patch checks the validity of the connection before attempting to send the device state. (closes issue ASTERISK-18078) Reported by: Michael L. Young Patches: res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young Tested by: Jonathan Rose ........ 2011-08-25 15:29 +0000 [r333203] Jason Parker * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac, makeopts.in, sounds/Makefile: Merged revisions 333201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines Fix installation into directories containing spaces. This also vastly simplifies the logic in sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1379/ ........ 2011-08-24 16:51 +0000 [r333115] Matthew Nicholson * res/res_fax.c: Changed the "timeout" option to "gwtimeout". ASTERISK-18219 2011-08-23 18:15 +0000 [r332878-333011] Richard Mudgett * /, apps/app_queue.c: Merged revisions 333010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) | 12 lines Memory Leak in app_queue The patch that was committed in the 1.6.x versions of Asterisk for ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8 but the other is. queue_leak.patch fixes the portion applicable to 1.8. (closes issue ASTERISK-18265) Reported by: Fred Schroeder Patches: queue_leak.patch (license #5049) patch uploaded by mmichelson Tested by: Thomas Arimont ........ * /, main/config.c: Merged revisions 332939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) | 7 lines Minor code optimizations. * Simplify ast_category_browse() logic for easier understanding. * Remove dead code in ast_variable_delete() and simplify some of its logic. ........ * /, apps/app_queue.c: Merged revisions 332874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) | 18 lines Reference leaks in app_queue. * Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. ........ 2011-08-22 19:43 +0000 [r332877] Paul Belanger * /, channels/chan_gtalk.c: Merged revisions 332876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) | 6 lines Revert previous commit It seems google is still making changes to the protocol. (issue ASTERISK-18301) ........ 2011-08-22 19:41 +0000 [r332875] Richard Mudgett * /: Fix merge property. 2011-08-22 18:40 +0000 [r332832] Matthew Jordan * apps/app_voicemail.c, include/asterisk/test.h, main/manager.c, /, main/file.c, main/test.c, main/app.c, configs/manager.conf.sample, include/asterisk/manager.h: Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ 2011-08-22 18:32 +0000 [r332761-332830] Richard Mudgett * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 332816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines Memory leaks in realtime_multi_xxx() when database access returns error. * Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. ........ * /, main/config.c: Merged revisions 332759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) | 15 lines Memory leak reading realtime database variable list. Calling ast_load_realtime() can leak the last list node if the read list only contains empty variable value items. * Fixed list filter loop in ast_load_realtime() to delete the list node immediately instead of the next time through the loop. The next time through the loop may not happen if the node to delete is the last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265) Patches: jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett ........ 2011-08-22 16:29 +0000 [r332756] Matthew Nicholson * res/res_fax.c, include/asterisk/res_fax.h: add a way to disable and/or modify the gateway timeout ASTERISK-18219 2011-08-21 14:33 +0000 [r332700] Paul Belanger * /, channels/chan_gtalk.c: Merged revisions 332699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue ASTERISK-18301) Reported by: az1324 ........ 2011-08-19 19:59 +0000 [r332654] Kinsey Moore * apps/app_confbridge.c: Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO doesn't behave as well in edge cases as MEETME_INFO. With this patch, CONFBRIDGE_INFO should behave in a much more reasonable manner when presented with invalid conferences and keywords. Review: https://reviewboard.asterisk.org/r/1359/ 2011-08-18 21:34 +0000 [r332560] Terry Wilson * main/netsock2.c, /: Merged revisions 332559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 5 lines Fix possible error on stringification of IPv4-mapped addrs The FreeBSD netsock2 test has been failing for a while. We were pasing sa->len to getnameinfo instead of sa_tmp->len. ASTERISK-18289 ........ 2011-08-18 19:29 +0000 [r332504] Kinsey Moore * channels/chan_dahdi.c, /: Merged revisions 332503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ 2011-08-18 14:48 +0000 [r332369-332447] Tilghman Lesher * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged revisions 332446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development mode, as it's useful when DEBUG_THREADS is enabled. ........ * Makefile, agi/Makefile, utils/Makefile, /, configure, include/asterisk/autoconfig.h.in, configure.ac, Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged revisions 332355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 10 lines Re-add support for spaces in pathnames, including now spaces in DESTDIR. This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ ........ 2011-08-17 18:09 +0000 [r332321] Terry Wilson * /, res/res_timing_timerfd.c: Merged revisions 332320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) | 10 lines Don't read from a disarmed or invalid timerfd Numerous isues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. This patch adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others. Review: https://reviewboard.asterisk.org/r/1361/ ........ 2011-08-17 16:01 +0000 [r332265] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ 2011-08-16 20:11 +0000 [r332177] Paul Belanger * tests/test_amihooks.c, tests/test_substitution.c, tests/test_heap.c, /, tests/test_expr.c, tests/test_ast_format_str_reduce.c, tests/test_logger.c, tests/test_gosub.c, tests/test_app.c, tests/test_dlinklists.c, tests/test_event.c, tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c, tests/test_netsock2.c, tests/test_pbx.c, tests/test_strings.c, tests/test_func_file.c, tests/test_security_events.c, tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c, tests/test_acl.c, tests/test_locale.c, tests/test_utils.c, tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c, tests/test_poll.c: Merged revisions 332176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug 2011) | 4 lines Flag test modules as 'core' Review: https://reviewboard.asterisk.org/r/1369/ ........ 2011-08-16 17:45 +0000 [r332119] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ 2011-08-16 17:17 +0000 [r332101] Richard Mudgett * /, main/features.c, CHANGES, configs/features.conf.sample, main/asterisk.c: Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ 2011-08-16 15:20 +0000 [r332022-332042] Matthew Nicholson * channels/sip/include/sip.h: fix a code comment AST-580 * UPGRADE.txt, CHANGES: Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ 2011-08-15 17:35 +0000 [r331956] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ 2011-08-15 15:22 +0000 [r331894] Paul Belanger * main/rtp_engine.c, /: Merged revisions 331886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug 2011) | 5 lines Fix noisy message when briding channels (closes issue ASTERISK-18270) Reported by: Federico Alves ........ 2011-08-15 15:14 +0000 [r331868] David Vossel * /, channels/chan_sip.c: Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) | 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ 2011-08-12 19:03 +0000 [r331775] Matthew Nicholson * /, apps/app_queue.c: Merged revisions 331774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug 2011) | 11 lines Unlock the channel before calling update_queue. Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock ........ 2011-08-12 18:59 +0000 [r331715-331772] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ * channels/chan_dahdi.c, /: Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ 2011-08-12 16:31 +0000 [r331659] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 331658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011) | 4 lines Fix netsock2 multiple zero-expansion test Remove erroneous single bracket. ........ 2011-08-12 16:21 +0000 [r331654] Kinsey Moore * /, main/logger.c: Merged revisions 331649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) | 12 lines Logger does not warn of failure to open logging channels Currently, logger only prints an error message to stderr when it fails to open a logger channel where many users will not see it because the logger lock is held. The alternative provided by this patch is to log the error to all attached consoles in the hopes that it will be easier to see. Additionally, this patch prevents the failed logger channel from being added to the list where it would silently fail on each call to the Asterisk logger. (closes issue ASTERISK-16231) Review: https://reviewboard.asterisk.org/r/1338 ........ 2011-08-12 16:18 +0000 [r331644] Jonathan Rose * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme ........ 2011-08-11 21:54 +0000 [r331579] Jason Parker * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) ........ 2011-08-11 21:42 +0000 [r331576] Richard Mudgett * /, funcs/func_shell.c: Merged revisions 331575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines Segfault in shell_helper in func_shell.c. The return value of popen() was not checked for failure to open. (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles Tested by: rmudgett ........ 2011-08-10 22:23 +0000 [r331518] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ 2011-08-10 20:41 +0000 [r331418-331462] Richard Mudgett * /, main/logger.c: Merged revisions 331461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) | 30 lines Output of queue log not started until logger reloaded. ASTERISK-15863 caused a regression with queue logging. The output of the queue log is not started until the logger configuration is reloaded. * Queue log initialization is completely delayed until the first message is posted to the queue log system. Including the initial opening of the queue log file. * Fixed rotate_file() ROTATE strategy to give the file just rotated out to the configured exec function after rotate. Just like the other strategies. * Fixed logger reload to always post the queue reload entry instead of just if there is a queue log file. * Refactored some code to eliminate some redundancy and to reduce stack utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett (closes issue ASTERISK-18208) Reported by: Christian Pinedo Review: https://reviewboard.asterisk.org/r/1333/ ........ * main/features.c: Make sure feature_request_and_dial() initializes outstate if passed in. * main/features.c, CHANGES: Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) 2011-08-10 13:48 +0000 [r331316] Kinsey Moore * main/manager.c, /: Merged revisions 331315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | 8 lines AMI action ModuleReload returns Error if Module: missing or empty An empty string was not being checked for properly causing identification of the module to be reloaded to fail and return an Error with message "No such module." (closes issue AST-616) ........ 2011-08-09 23:12 +0000 [r331265] Richard Mudgett * apps/app_parkandannounce.c, main/pbx.c, /, channels/chan_sip.c, main/features.c, channels/chan_iax2.c: Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ 2011-08-09 16:36 +0000 [r331147-331200] Alexandr Anikin * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c: Setup IP proto version for call in GK mode Added additional check for IP semantics before parse destination by ast_parse_args due to it can parse numeric as IP. (closes issue ASTERISK-18218) Reported by: slesru Patch: ASTERISK-18218.patch * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /, addons/ooh323c/src/ooLogChan.c: Merged revisions 331146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines move ast_cond_signal for admitted call after all data filled/freed clear all log channels by pointed number not only first free allocated callToken in ooh323_answer ........ 2011-08-09 15:59 +0000 [r331138-331143] Jason Parker * /, doc/asterisk.8: Merged revisions 331142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line Regenerate asterisk man page from sgml. ........ * doc/asterisk.sgml, /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample: Merged revisions 306999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir ........ * doc/asterisk.sgml, /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample: Revert merge of r306999, due to merge conflict. 2011-08-08 22:59 +0000 [r331041-331097] Terry Wilson * UPGRADE.txt, CHANGES, include/asterisk/manager.h: Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. * main/channel.c, CHANGES: Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issue ASTERISK-17455) 2011-08-08 20:53 +0000 [r331039] Kinsey Moore * /, res/res_musiconhold.c: Merged revisions 331038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines In-queue MOH stops after a periodic announcement If the seek value is past the end of file when resuming G.722 MOH, MOH will cease to function for the duration of the MOH session through all starts and stops until saved state is cleared. Adjusting the code to guarantee a single valid read (which is already assumed) fixes the bug. (closes issue ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/ Tested-by: Jonathan Rose ........ 2011-08-05 15:53 +0000 [r330940] David Vossel * codecs/codec_resample.c: The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly. 2011-08-05 07:38 +0000 [r330899] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, /, addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines change gk client behaivour on rrq/grq failures to setup timers and next tries after timeout instead of complete failure in the ooh323 stack ........ 2011-08-04 20:51 +0000 [r330844] Terry Wilson * /, configure, configure.ac: Merged revisions 330843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011) | 4 lines Make libsrtp instructions more explicit when linking fails (closes issue ASTERISK-18139) ........ 2011-08-03 15:15 +0000 [r330706-330763] Kinsey Moore * /, main/Makefile: Merged revisions 330762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) | 9 lines editing files in main/editline does not ensure rebuild of libedit.a When editing a source file in main/editline, the build system does not rebuild libedit.a and uses the already existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 330705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ 2011-08-02 20:52 +0000 [r330649] Kevin P. Fleming * /, res/res_jabber.c: Merged revisions 330648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 Aug 2011) | 2 lines Convert an error message to actually be helpful. ........ 2011-08-02 16:17 +0000 [r330576-330586] David Vossel * /, channels/chan_iax2.c: Merged revisions 330581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 resulting from an edge case in the way control frames are queued during calltoken negotiation is complete. (closes issue ASTERISK-17610) Reported by: mgrobecker ........ * /, channels/chan_sip.c: Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) | 2 lines Optimization to buffer initialization fix. ........ * /, channels/chan_sip.c: Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) | 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ 2011-08-01 15:23 +0000 [r330434] Kinsey Moore * /, main/say.c: Merged revisions 330433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | 9 lines Incorrect playback for Spanish in some circumstances When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you must use female pronunciation "1F". The function "say_date_with_format_es" does not take this in account. (closes ASTERISK-15016) Patch-by: Luis Jimenez ........ 2011-07-30 23:57 +0000 [r330369] Richard Mudgett * main/channel.c, /: Merged revisions 330368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines Remove some redundant locking code in ast_do_masquerade(). Also updated some comments. ........ 2011-07-30 15:34 +0000 [r330312] Gregory Nietsky * main/channel.c, /: Merged revisions 330311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines prevent double masqurading channels when one is been hung up and deadlock avoidance is used. There is a race condition in ast_do_masquerade / ast_hangup (at least) Reported by me signed off by schmidts with input from David Vossel Review: https://reviewboard.asterisk.org/r/1323/ ........ 2011-07-29 17:19 +0000 [r330204-330217] Sean Bright * /, formats/format_wav.c: Merged revisions 330213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........ * /, formats/format_wav.c: Merged revisions 330203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) | 2 lines Only write to wav files that were opened to be written to. ........ 2011-07-29 05:25 +0000 [r330162] Paul Belanger * apps/app_confbridge.c: Fix typo pointed out on #asterisk Thanks notten 2011-07-28 21:44 +0000 [r330108] Terry Wilson * main/term.c, /: Merged revisions 330107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011) | 2 lines Make console colors work for TERM=xterm-256color ........ 2011-07-28 17:10 +0000 [r330051] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ 2011-07-28 15:45 +0000 [r329995] Jason Parker * /, channels/chan_sip.c: Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ 2011-07-28 15:28 +0000 [r329992] Matthew Nicholson * /, res/res_fax.c: Merged revisions 329991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul 2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading the res_fax config file Patch by: tzafrir Reported by: tzafrir (closes issue ASTERISK-18161) ........ 2011-07-28 13:03 +0000 [r329771-329952] Sean Bright * configs/confbridge.conf.sample: The default conf-usermenu says that '8' can be used to leave the conference, so put that in the sample user menu. '5' is supposed to extend the conference, but there doesn't appear to be a concept of that in the menu actions. * apps/app_confbridge.c: Correct the spelling of 'conference.' * /, channels/chan_sip.c: Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ * /, Makefile.moddir_rules: Merged revisions 329767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul 2011) | 8 lines Explicitly sort the module list so that the menuselect lists are sorted. (closes issue ASTERISK-18141) Reported by: Richard Miller Patches: sort-order.diff uploaded by seanbright (License #5060) Tested by: leifmadsen ........ 2011-07-27 18:11 +0000 [r329710] Jonathan Rose * /, configs/indications.conf.sample: Merged revisions 329709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) | 8 lines Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf (closes issue ASTERISK-16263) Reported by: richardf Patches: nz-indications.patch uploaded by richardf (License #6015) ........ 2011-07-27 15:25 +0000 [r329670] Sean Bright * main/loader.c: Sort the module list so that 'module show' is alphabetical. 2011-07-27 04:25 +0000 [r329614] Tilghman Lesher * /, cdr/cdr_odbc.c: Merged revisions 329613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011) | 6 lines Duration and billsec are swapped in high resolution time. Closes ASTERISK-18024 Patches: 20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003) ........ 2011-07-26 14:19 +0000 [r329528-329538] Jonathan Rose * apps/app_voicemail.c, /: Merged revisions 329529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | 5 lines Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more appropriate anyway. ........ * apps/app_voicemail.c, include/asterisk/app.h, /, configs/voicemail.conf.sample, main/app.c: Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ 2011-07-25 19:55 +0000 [r329472] Paul Belanger * /, main/enum.c: Merged revisions 329471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul 2011) | 2 lines Decrease verbose messages to debug, to help clean up CLI. ........ 2011-07-25 14:06 +0000 [r329430-329431] Gregory Nietsky * main/dsp.c, include/asterisk/dsp.h: dsp_process was enhanced to work with alaw and ulaw in addition to slin. noticed that some functions could be refactored here it is. Reported by: irroot Tested by: irroot, mnicholson Review: https://reviewboard.asterisk.org/r/1304/ * channels/chan_sip.c, channels/sip/include/sip.h: Remove lastmsgssent from sip it has not been working since 1.6 Clean up the return values to be consistant not currently used Add doxygen returns MWI Event is sent on Register (closes issue ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak Review: https://reviewboard.asterisk.org/r/1272/ 2011-07-22 21:14 +0000 [r329331-329334] Richard Mudgett * main/pbx.c, /: Make use less redundant loop construct for iterating over hints. * main/pbx.c, /: Merged revisions 329299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines Deadlocks dealing with dialplan hints during reload. There are two remaining different deadlocks reported dealing with dialplan hints. The deadlock in ASTERISK-17666 is caused by invalid locking order in ast_remove_hint(). The hints container must be locked before the hint object. The deadlock in ASTERISK-17760 is caused by a catch-22 situation in handle_statechange(). The deadlock is caused by not having the conlock before calling the watcher callbacks. Unfortunately, having that lock causes a different deadlock as reported in ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made handle_statechange() no longer call the watcher callbacks holding any locks that matter. * Made hint ao2 destructor do the watcher callbacks for extension deactivation to guarantee that they get called. * Fixed hint reference leak in ast_add_hint() if the callback container constructor failed. * Fixed hint reference leak in complete_core_show_hint() for every hint it found for CLI tab completion. * Adjusted locking in ast_merge_contexts_and_delete() for safety. * Added context_merge_lock to prevent ast_merge_contexts_and_delete() and handle_statechange() from interfering with each other. * Fixed ast_change_hint() not taking into account that the extension is used for the hash key. (closes issue ASTERISK-17666) Reported by: irroot Tested by: irroot JIRA SWP-3318 (closes issue ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/ ........ 2011-07-22 Leif Madsen * Asterisk 10.0.0-beta1 Released. 2011-07-21 20:22 +0000 [r329257] Russell Bryant * channels/chan_dahdi.c, main/features.c, include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c, include/asterisk/rtp_engine.h: s/1.10/10.0/ 2011-07-21 18:05 +0000 [r329200-329204] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ * /, apps/app_directed_pickup.c: Merged revisions 329199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-17494 for v1.8 and later. (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith Tested by: Erik Smith ........ 2011-07-21 17:27 +0000 [r329188] Jason Parker * UPGRADE.txt: Fix version number in UPGRADE.txt. 2011-07-21 16:52 +0000 [r329145] Richard Mudgett * /, main/features.c: Merged revisions 329144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked! This appears to be a leftover from when ast_channel was converted to ao2 objects. Simply removed the extraneous unlock. (closes issue ASTERISK-17772) ........ 2011-07-21 16:04 +0000 [r329104] Russell Bryant * / (added): Change Asterisk 2.0 to 2.0 in binary 2011-07-20 21:31 +0000 [r329056] Paul Belanger * /, UPGRADE-1.8.txt: Merged revisions 329055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for PRI support. ........ ................ 2011-07-20 20:19 +0000 [r328996] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 328992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines Merged revisions 328987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines We can't guarantee an eth0 is present FreeBSD test fails on this case presumably because there is no eth0 on the test machine. Better to just remove this test for now. ........ ................ 2011-07-20 19:03 +0000 [r328937] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 328936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ ................ 2011-07-19 21:32 +0000 [r328880-328881] Kevin P. Fleming * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged revisions 328879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines Merged revisions 328878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines Revert partial attempt at handling pathnames with spaces. Revision 299794 attempted to improve the build system to be able to handle pathnames (primarily DESTDIR) with spaces in them, since this is common on some platforms (including Mac OSX). Unfortunately, the changes were incomplete and did not actually provide the desired behavior, and as a side effect the functionality that ensured that stale headers in the Asterisk 'include' directory were removed got broken. In addition, the check for stale (and possibly incompatible) modules in the Asterisk 'modules' directory also got broken, and would never report any stale modules. Users upgrading to this version or later versions would then see unexpected module load errors. Since there are few users who actually want to install Asterisk into paths that contain spaces, and a proper fix for the build system would take many hours, the best solution for now is to just revert the partial solution. ........ ................ * /: Edit the merge properties to match their names. 2011-07-19 21:21 +0000 [r328877] Russell Bryant * /: Fix properties after twilson's change so merges still work 2011-07-19 18:07 +0000 [r328772-328825] Kinsey Moore * res/res_rtp_asterisk.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 328824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ ................ * /, apps/app_meetme.c: Merged revisions 328771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN entry for joining a conference. (closes AST-601) Review: https://reviewboard.asterisk.org/r/1305/ ........ ................ 2011-07-19 02:00 +0000 [r328718] Terry Wilson * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c (added): Merged revisions 328717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This commit also adds linked list unit tests. Review: https://reviewboard.asterisk.org/r/1321/ ........ ................ 2011-07-18 20:51 +0000 [r328610-328665] Mark Murawki * apps/app_dial.c, /: Merged revisions 328664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines app_dial may double free a channel datastore When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it. (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ * /, channels/chan_sip.c: Merged revisions 328611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ * /, main/asterisk.c: Merged revisions 328609 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines Fixed invalid read and null pointer deref on asterisk shutdown. In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash. (closes issue ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher ........ ................ 2011-07-18 07:12 +0000 [r328542] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 328541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328541 | tilghman | 2011-07-18 02:11:26 -0500 (Mon, 18 Jul 2011) | 9 lines Merged revisions 328540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines Typo ........ ................ 2011-07-15 21:41 +0000 [r328502] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, /: Merged revisions 328428-328429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines Merged revisions 328427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines small gk processing fixes: - decrease for 1 second registration ttl for very low expirations (some providers expire few earlier than TTL) - delete rrq and registration expire timers on URQ received as we make new registration. ........ ................ r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines delete unproperly changed svn props ................ 2011-07-15 21:19 +0000 [r328449-328459] Leif Madsen * /, apps/app_macro.c: Merged revisions 328451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line Build app_macro by default because things depend on it. ........ * /, UPGRADE-1.10.txt, UPGRADE.txt, CHANGES: Merged revisions 328448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines Update UPGRADE.txt and CHANGES files. Update documentation files stating that deprecated modules are no longer built by default. ........ 2011-07-15 08:19 +0000 [r328381] Damien Wedhorn * channels/chan_skinny.c: Add SLA to skinny. Adds sublines to skinny lines. Each subline can be attached to an SLA station/trunk combo. Includes the following functionality: Callid is persistent for both in/out calls on all skinny devices. Can join, hold, resume. All sublines appear under a single line button. See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc. (closes issue ASTERISK-17947) Review: https://reviewboard.asterisk.org/r/1239/ 2011-07-15 00:23 +0000 [r328318-328344] Richard Mudgett * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/extconf.h, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 328329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ * /, channels/chan_sip.c: Merged revisions 328317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines Merged revisions 328302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........ ................ 2011-07-14 20:28 +0000 [r328259] Leif Madsen * funcs/func_speex.c, apps/app_playtones.c, bridges/bridge_softmix.c, apps/app_alarmreceiver.c, res/res_calendar_caldav.c, apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c, res/res_pktccops.c, channels/chan_skinny.c, pbx/pbx_ael.c, formats/format_h263.c, cdr/cdr_odbc.c, cdr/cdr_manager.c, utils/refcounter.c, funcs/func_timeout.c, formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, bridges/bridge_simple.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_db.c, funcs/func_callcompletion.c, funcs/func_channel.c, funcs/func_iconv.c, pbx/pbx_config.c, res/res_odbc.c, apps/app_voicemail.c, formats/format_sln.c, apps/app_authenticate.c, apps/app_readexten.c, res/res_phoneprov.c, apps/app_userevent.c, codecs/codec_gsm.c, tests/test_func_file.c, apps/app_setcallerid.c, res/res_config_odbc.c, funcs/func_audiohookinherit.c, apps/app_osplookup.c, funcs/func_odbc.c, cel/cel_custom.c, tests/test_utils.c, apps/app_mp3.c, res/res_timing_timerfd.c, codecs/codec_resample.c, formats/format_h264.c, apps/app_directory.c, formats/format_siren14.c, tests/test_amihooks.c, res/res_config_pgsql.c, funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c, apps/app_forkcdr.c, funcs/func_blacklist.c, apps/app_sms.c, formats/format_g723.c, utils/extconf.c, tests/test_poll.c, apps/app_stack.c, apps/app_verbose.c, utils/check_expr.c, funcs/func_module.c, codecs/codec_adpcm.c, tests/test_event.c, cdr/cdr_adaptive_odbc.c, apps/app_image.c, formats/format_wav_gsm.c, utils/stereorize.c, pbx/pbx_loopback.c, tests/test_time.c, funcs/func_shell.c, apps/app_skel.c, channels/chan_alsa.c, apps/app_externalivr.c, apps/app_milliwatt.c, formats/format_gsm.c, res/res_speech.c, apps/app_dial.c, apps/app_page.c, apps/app_fax.c, utils/astman.c, apps/app_disa.c, res/res_monitor.c, apps/app_waitforring.c, addons/cdr_mysql.c, res/res_fax_spandsp.c, res/res_timing_kqueue.c, apps/app_chanspy.c, apps/app_cdr.c, channels/chan_unistim.c, funcs/func_base64.c, channels/chan_multicast_rtp.c, funcs/func_md5.c, apps/app_meetme.c, tests/test_gosub.c, funcs/func_sysinfo.c, funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c, res/res_config_sqlite.c, apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c, channels/chan_phone.c, funcs/func_enum.c, main/manager.c, funcs/func_groupcount.c, tests/test_stringfields.c, tests/test_locale.c, tests/test_devicestate.c, funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c, tests/test_astobj2.c, apps/app_ivrdemo.c, res/res_clioriginate.c, apps/app_jack.c, apps/app_nbscat.c, res/res_calendar_icalendar.c, codecs/codec_a_mu.c, tests/test_ast_format_str_reduce.c, tests/test_dlinklists.c, res/res_convert.c, apps/app_waituntil.c, pbx/pbx_lua.c, utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c, cdr/cdr_tds.c, channels/chan_usbradio.c, apps/app_flash.c, apps/app_senddtmf.c, funcs/func_callerid.c, addons/app_saycountpl.c, cel/cel_pgsql.c, apps/app_dahdibarge.c, channels/chan_local.c, funcs/func_dialgroup.c, tests/test_logger.c, apps/app_record.c, funcs/func_env.c, funcs/func_strings.c, res/res_timing_dahdi.c, apps/app_chanisavail.c, bridges/bridge_multiplexed.c, res/res_rtp_multicast.c, cel/cel_odbc.c, channels/chan_dahdi.c, pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_pcm.c, apps/app_dumpchan.c, main/http.c, res/res_clialiases.c, res/res_calendar_exchange.c, res/res_ais.c, funcs/func_sprintf.c, codecs/codec_g722.c, tests/test_expr.c, cel/cel_tds.c, tests/test_app.c, utils/smsq.c, apps/app_morsecode.c, formats/format_ogg_vorbis.c, tests/test_sched.c, res/res_calendar_ews.c, apps/app_speech_utils.c, tests/test_acl.c, apps/app_sendtext.c, funcs/func_cdr.c, utils/hashtest2.c, utils/ael_main.c, apps/app_mixmonitor.c, formats/format_g726.c, utils/streamplayer.c, res/res_calendar.c, cel/cel_radius.c, channels/chan_vpb.cc, tests/test_heap.c, addons/format_mp3.c, res/res_snmp.c, apps/app_dictate.c, channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, res/res_jabber.c, funcs/func_uri.c, cel/cel_manager.c, apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, apps/app_read.c, cdr/cdr_syslog.c, codecs/codec_alaw.c, res/res_adsi.c, agi/eagi-test.c, utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c, formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c, pbx/pbx_dundi.c, funcs/func_devstate.c, addons/res_config_mysql.c, funcs/func_rand.c, apps/app_readfile.c, addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c, /, apps/app_sayunixtime.c, apps/app_test.c, res/res_http_post.c, res/res_smdi.c, main/features.c, funcs/func_srv.c, apps/app_amd.c, pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, formats/format_g719.c, channels/chan_bridge.c, apps/app_privacy.c, apps/app_echo.c, codecs/codec_speex.c, apps/app_saycounted.c, apps/app_dahdiras.c, channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c, apps/app_transfer.c, res/res_mutestream.c, apps/app_playback.c, res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c, tests/test_skel.c, apps/app_macro.c, apps/app_zapateller.c, codecs/codec_ilbc.c, addons/app_mysql.c, tests/test_substitution.c, utils/muted.c, utils/hashtest.c, funcs/func_sha1.c, formats/format_siren7.c, tests/test_security_events.c, funcs/func_config.c, bridges/bridge_builtin_features.c, funcs/func_volume.c, res/res_agi.c, apps/app_confbridge.c, addons/chan_mobile.c, apps/app_parkandannounce.c, res/res_security_log.c, cdr/cdr_custom.c, apps/app_while.c, res/res_rtp_asterisk.c, funcs/func_dialplan.c, funcs/func_db.c, apps/app_festival.c, res/res_limit.c, res/res_fax.c, apps/app_waitforsilence.c, channels/chan_console.c, apps/app_getcpeid.c, funcs/func_global.c, res/res_srtp.c, funcs/func_extstate.c, tests/test_strings.c, res/res_timing_pthread.c, apps/app_directed_pickup.c, channels/chan_h323.c, cel/cel_sqlite3_custom.c, codecs/codec_ulaw.c, channels/chan_nbs.c, formats/format_g729.c: Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ 2011-07-14 19:56 +0000 [r328208] Jonathan Rose * /, res/res_monitor.c: Merged revisions 328207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines Merged revisions 328205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines Monitor application arguments requirements fixed. Monitor was requiring options in spite of no individual option on Monitor being required. Review: https://reviewboard.asterisk.org/r/1320/ ........ ................ 2011-07-14 17:47 +0000 [r328163] Matthew Nicholson * /, main/dsp.c: Merged revisions 328162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines tune the v21 preamble detector to properly detect the desired sequence of hits and misses ........ 2011-07-13 22:10 +0000 [r328121] David Vossel * /, apps/app_mixmonitor.c: Merged revisions 328120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines Preserve sample rate quality of wideband mixmonitor recordings. MixMonitor has the ability to record in any file format Asterisk supports, but the quality of wideband audio is not preserved. This is because regardless of the sample rate the call is being recorded in, the audio is always downsampled to 8khz and then upsampled to whatever wideband format it is being written as. This patch resolves this by requesting the audio from the audiohook in the signed linear format closest to the sample rate of the format we are writing. This fix is only possible for Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband audio. Review: https://reviewboard.asterisk.org/r/1314/ ........ 2011-07-13 21:06 +0000 [r328079] Leif Madsen * BUGS, UPGRADE-1.10.txt (added), UPGRADE.txt: Add UPGRADE-1.10.txt file from UPGRADE.txt. 2011-07-13 20:40 +0000 [r328075-328076] Russell Bryant * /: set 1.10 merge properties * /: remove 1.8 merge properties 2011-07-13 18:47 +0000 [r328016] Richard Mudgett * /, configs/features.conf.sample: Merged revisions 328014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) | 1 line Add ATXFER_NULL_TECH note in features.conf.sample. ........ 2011-07-12 23:02 +0000 [r327953] Kevin P. Fleming * main/manager.c, /: Merged revisions 327950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines Correct double-free situation in manager output processing. The process_output() function calls ast_str_append() and xml_translate() on its 'out' parameter, which is a pointer to an ast_str buffer. If either of these functions need to reallocate the ast_str so it will have more space, they will free the existing buffer and allocate a new one, returning the address of the new one. However, because process_output only receives a pointer to the ast_str, not a pointer to its caller's variable holding the pointer, if the original ast_str is freed, the caller will not know, and will continue to use it (and later attempt to free it). (reported by jkroon on #asterisk-dev) ........ 2011-07-12 20:08 +0000 [r327891] Matthew Nicholson * /, apps/app_directory.c: Merged revisions 327890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) | 2 lines search in the current context for 'a' and 'o' instead of 'default' ........ 2011-07-12 19:39 +0000 [r327889] Jason Parker * Makefile, /: Merged revisions 327888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | 1 line Fix uninstall target, so that modules dir gets cleared again. ........ 2011-07-12 19:18 +0000 [r327856] Matthew Jordan * /, apps/app_voicemail.c: Merged revisions 327852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines Added additional checks for mailbox / password beginning with '*' character A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character. (closes issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/ ........ 2011-07-12 15:38 +0000 [r327794] Tilghman Lesher * tests/test_substitution.c, /: Merged revisions 327793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability. The problem with using 'echo -n' is that it is not portable. While BSD systems required that the '-n' option be removed and interpreted, System V required that all strings should be echoed with no interpretation of options. This fundamental difference of behavior means that it is never possible to use the '-n' flag to echo in tests which are meant to be portable. In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the System V semantics of the command, and thus the SHELL test failed on that platform. http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16 ........ 2011-07-12 15:23 +0000 [r327769] Matthew Nicholson * res/res_fax.c, include/asterisk/dsp.h, main/dsp.c: do v21 detection instead of CED detection for the fax gateway 2011-07-12 14:55 +0000 [r327749] David Vossel * main/bridging.c: Send video update frame to new video source in follow_talker correctly. 2011-07-12 14:40 +0000 [r327748] Kinsey Moore * apps/app_confbridge.c: Segfault on shutdown when confbridge is active When undergoing a shutdown and channels are kicked out of a bridge, a segfault occurs because ConfBridge tries to play sounds on the bridge after the underlying channels have been blown away due to the shutdown. (closes ASTERISK-18040) Review: https://reviewboard.asterisk.org/r/1283/ 2011-07-11 20:06 +0000 [r327684] Matthew Nicholson * tests/test_substitution.c: use printf instead of echo -n in substitution test 2011-07-11 19:49 +0000 [r327683] Terry Wilson * /, include/asterisk/jingle.h, channels/chan_gtalk.c: Merged revisions 327682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines Update chan_gtalk to work with changed GMail-based calls The messages sent by the GMail client have changed, but include the old-style messages as well. This patch checks for this case and uses the old-style offer. (closes issue ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ ........ 2011-07-11 18:44 +0000 [r327640] David Vossel * include/asterisk/bridging.h, bridges/bridge_softmix.c, main/bridging.c: Updates follow_talker video_mode in confbridge application. follow_talker mode originally echoed the same video stream to all participants. As the primary talker switched around, the video stream would result in the talker seeing themselves. Now the primary talker sees the last person who was talking rather than themselves. 2011-07-11 17:23 +0000 [r327469-327598] Matthew Nicholson * res/res_fax.c: renamed fax_gateway_send_ced() to fax_gateway_request_t38() * res/res_fax.c: actually do something with the ced timeout, also added more debug output * res/res_fax.c: write silence on the channel during t.38 negotiation * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul 2011) | 2 lines reset our buffer each iteration when doing variable substitution ........ * res/res_fax.c: Delay sending an CED tone generated T.38 reinvite to give the CED tone generating party time to send its own T.38 reinvite. Also don't forward frames through the gateway if we are negotiating T.38. * res/res_fax.c: fixed wording in a comment 2011-07-11 10:57 +0000 [r327413] Tzafrir Cohen * /, main/Makefile: Merged revisions 327411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | 5 lines fix building the Debian armhf (HardFloat) port Fixes http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385 (Missing pthreads) ........ 2011-07-10 01:37 +0000 [r327359] Alexandr Anikin * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample: Full T.38 handshaking and fax detection Add full t.38 handshaking for OOH323 that are required for newest T.38 gateway codes. Add fax detection (cng tone, t38) and dialplan redirection to fax ext on fax event detected. Add OOH323() function to set/get t38support and faxdetect parameters. (closes issue ASTERISK-17754) Reported by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot (license 52) issue19183-final.patch uploaded by may213 (license 454) Tested by: may213, irroot Review: https://reviewboard.asterisk.org/r/1174/ 2011-07-08 22:25 +0000 [r327246] Jason Parker * main/stdtime, utils, codecs, utils/db1-ast/recno, apps, cel, apps/confbridge, cdr, formats, codecs/gsm/src, utils/db1-ast/hash, funcs, bridges, codecs/lpc10, utils/db1-ast/db, codecs/g722, utils/db1-ast/mpool, main, codecs/speex, channels/sip, pbx, res, res/ael, channels, utils/db1-ast/btree: Add .o files to svn:ignore property, since it's only ignored if locally configured to do so. 2011-07-08 21:43 +0000 [r327212] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 327211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines INVITE 403 Forbidden response always retransmits the maximum times. Asterisk sends a 403 Forbidden response if authentication fails for an INVITE as required. However, it ignores the ACK and keeps retransmitting the response. * Made not delete the to-tag in the dialog so the expected ACK can be matched with the dialog and stop the retransmissions. ........ 2011-07-08 20:33 +0000 [r327116-327168] David Vossel * UPGRADE.txt, CHANGES: Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo in CHANGES as well. * CHANGES: Updates CHANGES log to reflect new slinear read/write file interpreters. * formats/format_sln.c, formats/format_sln16.c (removed): Support for writing and reading raw slin files 8khz-192khz. * formats/format_attr_silk.c (removed), formats/format_attr_celt.c (removed), res/res_format_attr_silk.c (added), res/res_format_attr_celt.c (added): Moves celt and silk format attribute files into res folder. It was inconsistent to have the silk and celt format attribute modules in the format file interpreter folder. 2011-07-08 19:54 +0000 [r327107] Matthew Nicholson * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines Reset our ast_str before passing it on to dialplan function backends. It is possible for a dialplan backend to not modify the given buffer or ast_str and still return success. This causes any previous value stored in the buffer to be used as if the new function call provided it. Some functions also append to the given buffer assuming it is empty. The test_substitution unit test has also been modified to detect this problem. (closes issue ASTERISK-17878) ........ 2011-07-08 16:00 +0000 [r327045-327047] Russell Bryant * /, tests/test_netsock2.c: Merged revisions 327046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08 Jul 2011) | 2 lines Fix an error and add more log message info to help see why this fails on FreeBSD. ........ * channels/chan_dahdi.c, /: Merged revisions 327044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) | 2 lines Resolve some set-but-unused-variable warnings. ........ 2011-07-08 01:26 +0000 [r327000] Richard Mudgett * main/pbx.c, /: Merged revisions 326985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines Some code cleanup in pbx.c * Mostly comment and format changes. * ast_context_remove_extension_callerid() and ast_add_extension_nolock() will write lock the found specific context. * ast_context_find() will now tolerate a NULL name. * Eliminated some inlined versions of find_context() and find_context_locked(). ........ 2011-07-07 22:39 +0000 [r326943] Jason Parker * include/asterisk/celt.h: I think reviewboard broke this. The whole file was doubled. 2011-07-07 22:17 +0000 [r326855-326904] David Vossel * formats/format_attr_celt.c (added): Adds the format_attr_celt file which was also missing from the CELT pass through patch. * include/asterisk/celt.h (added): Adds missing celt.h file from celt pass-through support patch. * CHANGES: Fixes spelling errors in CHANGES as well as adding a few entries for CELT and confbridge. * main/channel.c, main/format.c, res/res_rtp_asterisk.c, main/frame.c, main/rtp_engine.c, channels/chan_sip.c, include/asterisk/format.h, configs/codecs.conf.sample: Adds pass-through support for codec CELT. This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ 2011-07-07 19:20 +0000 [r326842] Tilghman Lesher * /, res/res_http_post.c: Merged revisions 326830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) | 1 line libgen.h is also needed on Darwin for basename(3) ........ 2011-07-07 17:24 +0000 [r326782] David Vossel * configs/confbridge.conf.sample, apps/confbridge/include/confbridge.h, apps/app_confbridge.c, apps/confbridge/conf_config_parser.c: Updates confbridge.conf video documentation and adds dtmf action for releasing video src. 2011-07-07 16:50 +0000 [r326750] Terry Wilson * utils/astdb2sqlite3.c, main/db.c: Use older functions out of deference to older distros 2011-07-07 16:18 +0000 [r326694] Jonathan Rose * res/res_config_odbc.c, /: Merged revisions 326689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines res_odbc patch by tilghman to fix integers with null values Addresses some improper sql statements in res_odbc that would cause an update to fail on realtime peers due to trying to set as "(NULL)" rather than an actual NULL. (closes issue #1922STERISK-17791) Reported by: marcelloceschia Patches: 20110505__issue19223.diff.txt uploaded by tilghman (license 14) ........ 2011-07-07 15:28 +0000 [r326682-326684] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 326683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) | 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ * /, channels/chan_sip.c: Merged revisions 326681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) | 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ 2011-07-07 09:49 +0000 [r326636] Tzafrir Cohen * contrib/scripts/live_ast: live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ 2011-07-06 20:58 +0000 [r326589] Terry Wilson * utils/db1-ast/btree/bt_open.c, utils/db1-ast/hash/hash_log2.c, utils/db1-ast/hash/hsearch.c, utils/db1-ast/btree/bt_page.c, utils/db1-ast/hash/page.h, utils/db1-ast/mpool, configure, utils/db1-ast/btree/extern.h, utils/db1-ast/include/db.h, main/db.c, utils/db1-ast/btree/bt_seq.c, utils/db1-ast/recno/recno.h, main/Makefile, utils/db1-ast/btree/bt_utils.c, utils/db1-ast/recno/rec_seq.c, configure.ac, utils/db1-ast/btree/bt_close.c, CHANGES, utils/db1-ast/hash/search.h, utils/db1-ast/hash/README, utils/db1-ast/recno/rec_open.c, utils/db1-ast/hash/hash_bigkey.c, utils/db1-ast/recno/rec_delete.c, Makefile, utils/db1-ast/include, utils/db1-ast/hash/hash_buf.c, utils/db1-ast/db, utils/db1-ast/libdb.map, utils/db1-ast/include/ndbm.h, utils/db1-ast/include/compat.h, utils/db1-ast/mpool/mpool.c, utils/db1-ast/btree/bt_debug.c, main/asterisk.c, utils/db1-ast (added), utils/db1-ast/btree/bt_split.c, utils, utils/db1-ast/recno, utils/db1-ast/btree/bt_delete.c, utils/db1-ast/include/circ-queue.h, tests/test_db.c, utils/db1-ast/Makefile, utils/db1-ast/hash/extern.h, utils/db1-ast/recno/rec_search.c, utils/db1-ast/btree/bt_get.c, utils/db1-ast/hash/hash.c, utils/db1-ast/btree/btree.h, utils/db1-ast/db/db.c, utils/db1-ast/hash/hash.h, utils/db1-ast/include/mpool.h, utils/db1-ast/recno/rec_get.c, utils/db1-ast/hash/hash_func.c, utils/utils.xml, utils/astdb2sqlite3.c (added), utils/db1-ast/btree/bt_overflow.c, UPGRADE.txt, utils/db1-ast/btree/bt_conv.c, utils/db1-ast/btree/bt_search.c, utils/db1-ast/btree/bt_put.c, utils/db1-ast/recno/rec_utils.c, utils/Makefile, utils/db1-ast/hash/hash_page.c, utils/db1-ast/hash, utils/db1-ast/mpool/README, utils/db1-ast/hash/ndbm.c, main/db1-ast (removed), utils/db1-ast/recno/rec_close.c, utils/db1-ast/recno/rec_put.c, utils/db1-ast/recno/extern.h, utils/db1-ast/btree: Replace Berkeley DB with SQLite 3 There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. 2011-07-06 17:39 +0000 [r326485-326544] David Vossel * channels/chan_sip.c: Fixes newlines from being stripped from out of dialog sip MESSAGES. * /, res/res_timing_timerfd.c: Merged revisions 326484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011) | 10 lines Reverts fix for timerfd locking issue. jrose discovered a performance issue with this fix that prevents his analog phones from working when using timerfd as a timing source. Until it is understood what is causing this performance problem, this patch is being reverted. ........ 2011-07-05 22:11 +0000 [r326412] Tilghman Lesher * channels/chan_jingle.c, channels/chan_dahdi.c, funcs/func_speex.c, /, channels/chan_sip.c, codecs/codec_speex.c, funcs/func_aes.c, pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c, channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ 2011-07-05 20:25 +0000 [r326368] Kinsey Moore * contrib/scripts/file.convert.sh (added): Prompt conversion script Several variables in the script control which files are converted and the source and destination formats. Patch-by: Trey Blancher (closes AST-560) 2011-07-05 17:35 +0000 [r326321] Richard Mudgett * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 326291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ 2011-07-05 16:46 +0000 [r326267] Mark Murawki * main/manager.c, CHANGES: New feature: AMI Action FilterAdd This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session (closes issue ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski 2011-07-05 13:38 +0000 [r326210] Matthew Jordan * /, main/file.c: Merged revisions 326209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines Updated filestream destructor to block until move is complete when cache is used When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes. (closes issue ASTERISK-17724) Reported by: Adiren P. Tested by: mjordan ........ 2011-07-01 21:11 +0000 [r326145] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 326144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ 2011-07-01 16:36 +0000 [r326056-326101] Gregory Nietsky * CHANGES: Change CHANGES move the commits to the right place r296249 r318141 Application changes * CHANGES: Change CHANGES move the commits to the right place in the file missed in review 2011-07-01 12:45 +0000 [r326006] Matthew Nicholson * res/res_fax.c, res/res_fax_spandsp.c: updated irroots info for the authors section 2011-06-30 21:05 +0000 [r325937] David Vossel * channels/chan_bridge.c: Fixes warning message caused by confbridge playback chan not being answered. 2011-06-30 20:47 +0000 [r325936] Richard Mudgett * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 325935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ 2011-06-30 20:33 +0000 [r325931] David Vossel * configs/confbridge.conf.sample, apps/confbridge/include/confbridge.h, include/asterisk/bridging.h, include/asterisk/dsp.h, bridges/bridge_softmix.c, apps/app_confbridge.c, CHANGES, main/bridging.c, main/dsp.c, apps/app_voicemail.c, apps/confbridge/conf_config_parser.c: Video support for ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/ 2011-06-30 20:24 +0000 [r325900] Matthew Jordan * /, apps/app_voicemail.c: Merged revisions 325877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines Patched voicemail user option for emailbody / emailsubject Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan ........ 2011-06-30 19:31 +0000 [r325864] Jonathan Rose * /, res/res_musiconhold.c: Merged revisions 325821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines Fixes an issue with Music on Hold classes losing files in playlist when realtime is used. The bug occurs rather intermittently and I relied on the reporters to test the patch. After a sanity check and some testing, I'm giving it an OK. (closes issue ASTERISK-17875) Reported by: David Cunningham Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009) ........ 2011-06-30 18:22 +0000 [r325815-325816] Matthew Nicholson * res/res_fax.c, include/asterisk/res_fax.h, CHANGES, res/res_fax_spandsp.c: Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38 terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the dialplan. Big thanks to irroot for porting this code to use the framehooks api. * main/frame.c: copy all flags on asterisk frames instead of just the timing flag 2011-06-29 21:50 +0000 [r325741] Kinsey Moore * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 325740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ 2011-06-29 19:02 +0000 [r325674] David Vossel * /, res/res_timing_timerfd.c: Merged revisions 325673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 Jun 2011) | 6 lines Fixes timerfd locking issue. (closes ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz Review: https://reviewboard.asterisk.org/r/1255/ ........ 2011-06-29 18:18 +0000 [r325611-325616] Richard Mudgett * /, apps/app_queue.c: Merged revisions 325614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed error exit cleanup in app_queue.c copy_rules() and reload_queue_rules(). ........ * /, apps/app_queue.c: Merged revisions 325610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines Response to QueueRule manager command does not contain ActionID if it was specified. * Add ActionID support as documented for the QueueRule AMI action. * Remove documentation for ActionID with the Queues AMI action. The output does not follow normal AMI response output and there is no place to put an ActionID header. (closes issue AST-602) Reported by: Vlad Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 ........ 2011-06-29 16:19 +0000 [r325538-325547] Matthew Nicholson * main/channel.c, /: Merged revisions 325545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines make framehooks prevent native bridging (for real this time) ........ * apps/app_dial.c, main/rtp_engine.c, /: Merged revisions 325537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines don't do native/remote bridging if a framehook is active on the channel ........ 2011-06-29 06:39 +0000 [r325483] Gregory Nietsky * configs/queues.conf.sample, UPGRADE.txt, CHANGES, apps/app_queue.c: Commit "distrotech" app_queue changes to Trunk * Added general option negative_penalty_invalid default off. when set members are seen as invalid/logged out when there penalty is negative. for realtime members when set remove from queue will set penalty to -1. * Added queue option autopausedelay when autopause is enabled it will be delayed for this number of seconds since last successful call if there was no prior call the agent will be autopaused immediately. * Added member option ignorebusy this when set and ringinuse is not will allow per member control of multiple calls as ringinuse does for the Queue. - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. (closes issue ASTERISK-17421) (closes issue ASTERISK-17391) Reported by: irroot Tested by: irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ 2011-06-28 21:51 +0000 [r325417] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 325416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines Fix random misspelling noticed on asterisk-users. ........ 2011-06-28 20:32 +0000 [r325345] David Vossel * /, channels/chan_sip.c: Merged revisions 325339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ 2011-06-28 17:38 +0000 [r325213] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 325212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ 2011-06-28 16:04 +0000 [r325153] Jonathan Rose * /, res/res_musiconhold.c: Merged revisions 325152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun 2011) | 5 lines Fixes moh reload breaking custom mode moh classes when the config file is untouched (closes issue ASTERISK-17730) Reported by: sdolloff ........ 2011-06-28 15:34 +0000 [r325151] David Vossel * channels/chan_sip.c: Fixes issue with video and text not being reinvited correctly with directmedia If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. 2011-06-28 15:12 +0000 [r325092] Leif Madsen * /, build_tools/prep_tarball: Merged revisions 325091 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 Jun 2011) | 1 line Remove line from prep_tarball that kills mkrelease. ........ 2011-06-28 00:07 +0000 [r325046] Terry Wilson * channels/chan_sip.c: Don't forget to build the Via when sending MESSAGE 2011-06-27 16:32 +0000 [r324961] Tilghman Lesher * /, main/asterisk.c: Merged revisions 324955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines Save and restore errno from within signal handlers. This is recommended by the POSIX standard, as well as by the sigaction(2) manpage for various platforms that we support (e.g. Mac OS X). ........ 2011-06-27 15:38 +0000 [r324915] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 324914 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines When subscribing MWI to an unsolicited mailbox the first notification is incorrect. A remote peer subscribed to MWI with the unsolicited option and a local phone subscribed to the remote mailbox. The notify message-summary events are sent correctly except for the first one when subscribing, which will always be 0. This means the phone MWI indicator will be wrong until the mailbox read/unread count changes and the event is fired. Looks like this is a regression from ASTERISK-16149. * Fix the logic to check the cache and if allowed then fallback to manually counting mailbox messages. (closes issue ASTERISK-17997) Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA SWP-3551 ........ 2011-06-24 20:50 +0000 [r324850] Richard Mudgett * /, pbx/pbx_config.c: Merged revisions 324849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines Syntax errors in dialplan do not display the file name. When issuing the CLI command "dialplan reload" syntax errors and warnings are displayed on the console. The offending line number is displayed on the console, but the file name is not displayed. Errors caught in main/config.c do display the file name. (closes issue ASTERISK-17985) Reported by: ulogic Patches: pbx_config.patch uploaded by ulogic (License #5685) modified format Tested by: rmudgett JIRA SWP-3554 ........ 2011-06-24 16:50 +0000 [r324769] Jonathan Rose * include/asterisk/logger.h, /: Merged revisions 324768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines DTMF wasn't being logged on connected consoles when enabled in logger.conf Previously in order for DTMF to be logged in a connected console session, the user would have to do logger set channel DTMF on. This corrects that so that it is on by default. This issue was caused by an off by one error incurred by a logger level count of 6 in logger.h where it should have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........ 2011-06-23 18:56 +0000 [r324708-324709] Kinsey Moore * apps/app_confbridge.c: ConfBridge: redundant code cleanup There is no reason to clean up features twice. Review: https://reviewboard.asterisk.org/r/1279/ * /, channels/chan_sip.c: Merged revisions 324678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 ........ ................ 2011-06-23 18:31 +0000 [r324664-324689] David Vossel * /, channels/sip/reqresp_parser.c: Merged revisions 324685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines Fixes sip crash when calling remove_uri_parameters with NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: jaredmauch ........ * /, main/features.c, channels/chan_iax2.c, include/asterisk/frame.h: Merged revisions 324652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ 2011-06-23 03:16 +0000 [r324558] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 324557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines Remove tests for parsing address with invalid port getaddrinfo on OS X returns with EAI_NONAME error when passed a port greater than 65535. Linux throws no error, so remove the tests for now. ........ 2011-06-22 19:17 +0000 [r324495] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 324491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line Use correct variable for text SRTP media. ........ 2011-06-22 19:12 +0000 [r324487] Terry Wilson * main/netsock2.c, /, channels/chan_sip.c, include/asterisk/netsock2.h, tests/test_netsock2.c (added): Merged revisions 324484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ 2011-06-22 18:45 +0000 [r324480-324482] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 324481 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a reference leak in an error path in sip_msg_send(). ........ r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines Timout or error on INFO or MESSAGE transaction causes call to be lost. When exchanging INFO messages within a call, 4xx error causes the call to be disconnected although RFC 2976 explicitly states that such transactions do not modify the state of the dialog. When exchanging MESSAGE messages within a call, 4xx error causes the call to be disconnected. To provide least surprise, we should not disconnect the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428 Section 2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review: https://reviewboard.asterisk.org/r/1257/ Review: https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........ * /, channels/chan_sip.c: Merged revisions 324479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line Comments and whitespace in chan_sip.c ........ 2011-06-21 21:55 +0000 [r324365-324422] David Vossel * apps/app_confbridge.c: Fixes issue with channel write format being incorrectly restored when MOH is used in confbridge. * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 324364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines Fixes locking inversion issue in ast_async_goto() During this function we can not hold the "chan" lock while doing the masquerade, the explicit goto on the tmp chan, or the channel alloc. Instead we need to get the channel lock, store off information about the channel that we need, and then let the channel lock go for the remainder of the function. Review: https://reviewboard.asterisk.org/r/1275/ ........ 2011-06-21 16:06 +0000 [r324304] Kinsey Moore * apps/app_confbridge.c: ConfBridge does not handle hangup properly When playing back a prompt to a channel, confbridge neglects to check for hangup events causing lockup condititions for hangups that occur before actually joining the conference. This change ensures that the user is removed from the conference in the event of a premature hangup. Review: https://reviewboard.asterisk.org/r/1277/ 2011-06-21 15:49 +0000 [r324302] David Vossel * channels/chan_sip.c: Fixes issue with finding correct extension when message context is used. 2011-06-20 18:13 +0000 [r324242] Leif Madsen * /, configs/queuerules.conf.sample: Merged revisions 324241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc ........ 2011-06-20 17:34 +0000 [r324238] Terry Wilson * /, channels/chan_sip.c: Merged revisions 324237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines Ignore media offers with a port of 0 Section 5.1 of RFC3264 states: A port number of zero in the offer indicates that the stream is offered but MUST NOT be used. (closes issue ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch uploaded by jacco (license 1277) Tested by: jacco, twilson ........ 2011-06-17 18:52 +0000 [r324177-324179] Leif Madsen * main/manager.c, /: Merged revisions 324178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) | 2 lines Add Username and Secret fields to manager Login action. Pointed out by Vlad Povorozniuc ........ * /, apps/app_meetme.c: Merged revisions 324176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines Fix typo in documentation. Pointed out by Vlad Povorozniuc ........ 2011-06-17 18:23 +0000 [r324175] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 324174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines Add header string to libpri debug output. Add header string to libpri debug output so the libpri output can be found/extracted easier from huge debug trace files. ........ 2011-06-17 15:32 +0000 [r324131] Leif Madsen * main/pbx.c, /: Merged revisions 324115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) | 3 lines Fix grammar in documentation for Goto() and GotoIf() (closes issue ASTERISK-18023) Reported by: Tim Osman ........ 2011-06-16 22:49 +0000 [r324050] Terry Wilson * main/channel.c, channels/chan_local.c, /, channels/chan_sip.c, include/asterisk/channel.h: Merged revisions 324048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ 2011-06-16 18:13 +0000 [r323991] Richard Mudgett * /, tests/test_event.c: Merged revisions 323990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) | 5 lines The test_event unit test is occasionally failing. Wait for the special posted event to process before adding a new subscription. ........ 2011-06-16 15:59 +0000 [r323673-323933] Terry Wilson * Makefile, /: Merged revisions 323932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) | 4 lines Don't assume ASTDBDIR exists It most likely doesn't on FreeBSD ........ * /, tests/test_db.c: Merged revisions 323866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) | 2 lines Remove now-useless cast of ARRAY_LEN ........ * include/asterisk/utils.h, /: Merged revisions 323863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86 and x86_64 systems ........ * /, tests/test_db.c: Merged revisions 323859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) | 2 lines Fix more ARRAY_LEN format string issues ........ * /, main/features.c: Merged revisions 323754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ ................ ................ * /, tests/test_db.c: Merged revisions 323672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) | 5 lines Cast ARRAY_LEN to size_t for ast_logging 32-bit and 64-bit machines return different types for ARRAY_LEN(), so cast it before using in a format string. ........ 2011-06-15 16:49 +0000 [r323671] Richard Mudgett * /, tests/test_event.c, main/event.c: Merged revisions 323669-323670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines [regression] Voicemail MWI is no longer sent. When leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. If you restart Asterisk, everything comes up at that state correctly, but changes to the messages in voicemail causes the light to not be set appropriately. Very easy to reproduce. * Made ast_event_check_subscriber() return TRUE if there are ANY subscribers to an event type when there are no restricting ie values passed. This allows an event being queued to be queued. (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621) (closes issue ASTERISK-18019) ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines Add a test to the event unit tests to catch ASTERISK-18002. The new tests check to see if there are ANY subscribers to the event type when ast_event_check_subscriber() is not passed any specific ie values. (issue ASTERISK-18002) ........ 2011-06-15 16:19 +0000 [r323621] Jonathan Rose * res/res_config_pgsql.c, /: Merged revisions 323610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun 2011) | 7 lines Adds PQclear calls on result to various parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported by: byronclark Patches: pgsql_pqclear.patch uploaded by byronclark (license 1200) ........ 2011-06-15 15:33 +0000 [r323609] Sean Bright * main/manager.c, /: Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ 2011-06-15 13:45 +0000 [r323517] Kinsey Moore * apps/app_confbridge.c, CHANGES: CONFBRIDGE_INFO function to get conference data Added the CONFBRIDGE_INFO dialplan function to get information about a conference bridge including locked status and number of parties, admins, and marked users. Review: https://reviewboard.asterisk.org/r/1271/ 2011-06-15 00:51 +0000 [r323397-323457] Richard Mudgett * /, main/event.c: Merged revisions 323456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) | 1 line Add missing break in ast_event_get_cached(). ........ * main/netsock2.c, main/dnsmgr.c, /: Merged revisions 323392,323394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) | 6 lines Add more strict hostname checking to ast_dnsmgr_lookup(). Change suggested in review. Review: https://reviewboard.asterisk.org/r/1240/ ........ r323394 | rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines Made ast_sockaddr_split_hostport() port warning msgs more meaningful. ........ 2011-06-14 17:03 +0000 [r323374] Terry Wilson * res/res_rtp_asterisk.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 323370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ 2011-06-14 16:47 +0000 [r323372] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 323371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this was causing NAT=Yes to always use rport when present which was against 1.6.2 behavior and the check itself was redundant since the only way this segment of code could be reached was if RPORT_PRESENT was already evaluated as true earlier. (closes issue ASTERISK-17789) Reported by: byronclark Patches: use_sip_nat_force_rport.patch uploaded by byronclark (license 1200) ........ 2011-06-14 14:37 +0000 [r323325] David Vossel * channels/chan_sip.c: Store sip peer name as var data on a outofcall msg. 2011-06-13 20:44 +0000 [r323272] Kinsey Moore * apps/confbridge/conf_config_parser.c: Config inheritance doesn't work with ConfBridge() menu definitions Current behavior in ConfBridge menu definitions is that first definition takes precedence, even in templated situations. This change allows inheritance and overriding to work as expected so that the last definition takes precedence. (closes ASTERISK-17986) Review: https://reviewboard.asterisk.org/r/1267/ 2011-06-13 19:54 +0000 [r323214] Leif Madsen * main/channel.c, /: Merged revisions 323213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) | 6 lines Avoid dividing by zero with L() option to Dial() Reported by: nicolasom Patches: issue-17995.patch - nicolasom (License #5994) ........ 2011-06-13 19:43 +0000 [r323212] David Vossel * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Addition of "outofcall_message_context" sip.conf option. Review: https://reviewboard.asterisk.org/r/1265/ 2011-06-13 19:03 +0000 [r323155] Leif Madsen * /, res/res_agi.c: Merged revisions 323154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) | 6 lines Tweak documentation for AGI Hangup command. (closes issue ASTERISK-17999) Reported by: Ben Klang Patches: hangup-doc.diff - uploaded by Ben Klang (License #5876) ........ 2011-06-13 14:38 +0000 [r323106-323107] Kinsey Moore * apps/confbridge/include/confbridge.h, apps/app_confbridge.c: MOH for only user not working with ConfBridge This adds the playing_moh flag to the conference_bridge_user struct that signifies when MOH should be playing so code doesn't have to guess whether MOH is playing. This change also adds the necessary checking to ensure that MOH continues playing for a single user in a conference after the join sound is played when configured to do so. (closes ASTERISK-17988) Review: https://reviewboard.asterisk.org/r/1263/ * apps/app_confbridge.c: ConfBridge: Use of bridge or user profiles that don't exist Bridge and user profiles are not checked for existence before use. The lack of a fully formed bridge profile can cause a segfault when sounds are accessed. This change ensures that bridge and user profiles exist prior to usage attempts. Review: https://reviewboard.asterisk.org/r/1264/ 2011-06-10 19:22 +0000 [r323041] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 323040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson ........ 2011-06-10 15:30 +0000 [r322866-322982] Terry Wilson * /, main/db.c: Merged revisions 322981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines Avoid a DB1 infinite loop bug Explicity check the last entry in the DB and make sure that we don't iterate past it. Since there can be no duplicates, this just makes sure that we stop after matching the last key. This patch also refactors the code to get away from some code duplication. A previous patch added many astdb tests and this patch passed them. Review: https://reviewboard.asterisk.org/r/1259/ ........ * /, tests/test_db.c (added): Merged revisions 322923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 Jun 2011) | 2 lines Add some astdb unit tests ........ * /, include/asterisk/astdb.h: Merged revisions 322865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 Jun 2011) | 4 lines Correct ast_db_deltree documentation ast_db_deltree returns -1 on error, otherwise the number of deletions ........ 2011-06-09 17:43 +0000 [r322808] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 322807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ 2011-06-09 16:47 +0000 [r322750] Richard Mudgett * /, apps/app_directed_pickup.c, main/features.c, include/asterisk/features.h: Merged revisions 322749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ 2011-06-09 11:05 +0000 [r322544] Damien Wedhorn * channels/chan_skinny.c: Add autoanswer to skinny. Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. 2011-06-08 20:48 +0000 [r322426-322485] Richard Mudgett * /, apps/app_queue.c: Merged revisions 322484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines Ring all queue with more than 255 agents will cause crash. 1. Create a ring-all queue with 500 permanent agents. 2. Call it. 3. Asterisk will crash. The watchers array in app_queue.c has a hard limit of 255. Bounds checking is not done on this array. No sane person should put 255 people in a ring-all queue, but we should not crash anyway. * Added bounds checking to the watchers array. JIRA AST-464 JIRA SWP-2903 ........ * main/dnsmgr.c, /: Merged revisions 322425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines SRV lookup attempted for SIP peers listed as an IP address. Asterisk attempts to SRV lookup a host name even if the host name is an IP address. Regression introduced when IPv6 support was added. * Restored the check in ast_dnsmgr_lookup() to see if the given host name is an IP address. The IP address could be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ ........ 2011-06-08 11:38 +0000 [r322381] Damien Wedhorn * channels/chan_skinny.c: Remove skinny do_monitor and use ast_sched_start instead The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ 2011-06-08 06:45 +0000 [r322323] Gregory Nietsky * /, channels/chan_sip.c: Merged revisions 322322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ 2011-06-07 23:14 +0000 [r322284] Richard Mudgett * channels/chan_sip.c, include/asterisk/message.h: Correct some whitespace and a reference debug message. 2011-06-07 19:17 +0000 [r322244] Russell Bryant * res/res_jabber.c: Actually check the "sendtodialplan" option setting for xmpp. (closes issue ASTERISK-17978) Reported by: elguero Patches: stop_messages_going_to_dialplan.patch (license #5026) 2011-06-07 18:01 +0000 [r322190] Paul Belanger * configs/sip_notify.conf.sample, /: Merged revisions 322189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ 2011-06-06 19:39 +0000 [r322111-322128] Gregory Nietsky * apps/app_queue.c: Remove Unused Var Warning rt_handle_member_record * apps/app_queue.c: Refactor rt_handle_member_record Review: https://reviewboard.asterisk.org/r/1172 2011-06-06 19:15 +0000 [r322070] Jonathan Rose * include/asterisk/logger.h, /, main/asterisk.c: Merged revisions 322069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ 2011-06-03 22:15 +0000 [r321814-321927] Richard Mudgett * cel/cel_radius.c, /, cdr/cdr_radius.c: Merged revisions 321926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines Asterisk crash when unloading cdr_radius/cel_radius. The rc_openlog() API call is passed a string that is used by openlog() to format log messages. The openlog() does not copy the string it just keeps a pointer to it. When the module is unloaded, the string is gone from memory. Depending upon module load order and if the other module then has an error, a crash happens. * Pass rc_openlog() a strdup'd string with the understanding that there will be a small memory leak if the cdr_radius/cel_radius modules are unloaded. * Call rc_destroy() to free the rc handle memory when the module is unloaded. JIRA AST-483 JIRA SWP-3062 ........ * /, main/ccss.c: Merged revisions 321924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines Be more explicit for CCSS generic device state event subscription. Make CCSS generic device state event subscription specify the AST_EVENT_IE_STATE ie exists to be safe. ........ * /, tests/test_event.c, main/event.c: Merged revisions 321871 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines Event subscription fixes. Must commit the subscription fixes together with the integration subscription tests. The subscription fixes cause an erroneously passing test to fail. The new subscription tests detect errors without the subscription fixes. * Added missing event_names[] table entry. * Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to correctly detect if a subscriber exists for the proposed event. * Made match_ie_val() and match_sub_ie_val_to_event() check the buffer length for RAW payload types. * Fixed error handling memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), and ast_event_queue(). * Made ast_event_new() and ast_event_check_subscriber() better protect themselves from an invalid payload type. * Added container lock protection between removing old cache events and adding the new cached event in ast_event_queue_and_cache()/event_update_cache(). * Added new event subscription tests. ........ * include/asterisk/event.h, /, channels/chan_sip.c, main/event.c, channels/chan_iax2.c: Merged revisions 321812-321813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ 2011-06-03 18:25 +0000 [r321752] Russell Bryant * tests/test_astobj2.c, main/astobj2.c: Fix some astobj2 iterator breakage, add another unit test. Review: https://reviewboard.asterisk.org/r/1254/ 2011-06-03 13:18 +0000 [r321689] Leif Madsen * /, configs/queues.conf.sample: Merged revisions 321685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol ........ 2011-06-02 22:09 +0000 [r321617] Russell Bryant * channels/chan_sip.c: Fix message destination extension. Don't send all messages to 's'. Get the destination from the request URI. (Found using automated test cases). 2011-06-01 23:12 +0000 [r321548] Richard Mudgett * main/cdr.c, /: Merged revisions 321547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) | 1 line CDR comment tweaks. ........ 2011-06-01 21:31 +0000 [r321546] Russell Bryant * main/channel.c, channels/chan_sip.c, configs/jabber.conf.sample, include/asterisk/message.h (added), include/asterisk/jabber.h, include/asterisk/channel.h, configs/sip.conf.sample, include/asterisk/_private.h, CHANGES, res/res_jabber.c, main/message.c (added), channels/sip/include/sip.h, main/asterisk.c: Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ 2011-06-01 20:11 +0000 [r321538] Brett Bryant * /, apps/app_voicemail.c: Merged revisions 321537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines This patch fixes an issue with using the wrong voicemail folders with greetings. (closes issue #17871) Reported by: edhorton Patches: digium_bug_17871_2 uploaded by fhackenberger (license 592) Tested by: edhorton, fhackenberger ........ 2011-06-01 10:45 +0000 [r321529] Alexandr Anikin * addons/chan_ooh323.c, /, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/oochannels.c: Merged revisions 321528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines Fix double alerting, add forced alerting before answer Fix double alerting (it wasn't fixed here by issue #18542) Add forced alerting before connect (if it wasn't before) Try to send all packets from outgoing queue rather than one only Call goes into clearing state when disconnect command is received (closes issue #19361) Reported by: vmikhelson Patches: issue19361-3.patch uploaded by may213 (license 454) Tested by: vmikhelson ........ 2011-05-31 20:55 +0000 [r321518] Richard Mudgett * include/asterisk/acl.h, /, include/asterisk/dnsmgr.h: Merged revisions 321517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) | 1 line Update some comments. ........ 2011-05-31 19:01 +0000 [r321516] David Vossel * channels/chan_local.c, /: Merged revisions 321515 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines Chan_local locking cleanup. This patch removes all of the unnecessary deadlock avoidance loops that occur in chan_local. It also resolves an issue with a deadlock triggered by local channel optimizations. (issue #18028) Review: https://reviewboard.asterisk.org/r/1231/ ........ 2011-05-31 16:06 +0000 [r321512] Leif Madsen * /, channels/chan_sip.c: Merged revisions 321511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines Enhance NOTICE message to know who couldn't access the dialplan. (closes issue #19390) Reported by: lmadsen Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) Tested by: russell ........ 2011-05-28 00:29 +0000 [r321338-321445] Richard Mudgett * /, res/res_agi.c: Merged revisions 321436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) | 4 lines Some hagi launch cleanup. Inspired by issue 19256. This patch would also fix the crash. ........ * main/srv.c, /: Merged revisions 321392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines Crash when using hagi and no servers are available. When none of the servers returned by the SRV querey respond, asterisk crashes. The problem is that if the loop over all the SRV entries finishes then the srv_context has already been cleaned up. * Make ast_srv_cleanup() check to see if the context is already cleaned up. (closes issue #19256) Reported by: byronclark ........ * /, apps/app_privacy.c, UPGRADE.txt, CHANGES: Merged revisions 321337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ 2011-05-27 21:40 +0000 [r321334] Leif Madsen * /, main/features.c: Merged revisions 321333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) | 7 lines Allow parking lot hints and musicclass to be set. (closes issue #19378) Reported by: sboily_proformatique Patches: pf_parkinghint_music_fix uploaded by sboily proformatique (license 206) Tested by: russell ........ 2011-05-27 21:37 +0000 [r321331-321332] Richard Mudgett * UPGRADE.txt: Add note about PrivacyManager to UPGRADE.txt * /, apps/app_privacy.c, CHANGES: Merged revisions 321330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. The trunk(v1.10) version will remove the unused options position. (closes issue #19273) Reported by: mdavenport ........ 2011-05-27 16:35 +0000 [r321289] Jonathan Rose * /, channels/sip/reqresp_parser.c: Merged revisions 321273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson. ........ 2011-05-27 08:37 +0000 [r321212] Alec L Davis * /, main/features.c: Merged revisions 321211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines Fix *8 directed pickup locks system during pickupsound play out move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 threads trying to write audio to same channel. In addition fixes choppy audio beep in issue 19177. (issue #18654) (issue #19177) Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ ........ 2011-05-26 21:50 +0000 [r321101-321156] Mark Murawki * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged revisions 321155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic. Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails. (closes issue #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: [full review board URL with trailing slash] ........ * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 321100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null pointer Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables (closes issue #19346) Reported by: kobaz Patches: netsock2.patch uploaded by kobaz (license 834) Tested by: kobaz, Marquis ........ 2011-05-26 18:10 +0000 [r321045] Richard Mudgett * /, include/asterisk/netsock2.h: Merged revisions 321044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) | 1 line Update ast_sockaddr comment with an important note. ........ 2011-05-26 17:35 +0000 [r321043] Terry Wilson * main/rtp_engine.c, /: Merged revisions 321042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) | 6 lines Initialize stack-allocated ast_sockaddrs before use It is important to always initialize ast_sockaddrs before use--even if they are passed to ast_sockaddr_copy as the underlying storage could be bigger than what ends up being copied--leaving part of the data unitialized. ........ 2011-05-26 16:54 +0000 [r321003] Russell Bryant * /, channels/chan_alsa.c: Merged revisions 320947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 May 2011) | 2 lines Remove some variables that were set but unused. ........ 2011-05-26 15:55 +0000 [r320946] Terry Wilson * main/channel.c, main/utils.c, include/asterisk/stringfields.h: Use va_copy for stringfields The ast_string_field_build_va functions were written to take to separate va_lists to work around FreeBSD 4 not having va_copy defined. In the end, we don't support anything using gcc < 3 anyway because we use va_copy all over the place anyway. This patch just simplifies things by removing the second va_list function arguments in favor of va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This line, and those below, will be ignored-- M include/asterisk/stringfields.h M main/utils.c M main/channel.c 2011-05-25 22:28 +0000 [r320820-320884] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 320883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC request has incorrect To/From SIP headers. They are reversed and the dialog tags are the same when they should not be. If pedantic mode was disabled, then the cancel would have succeeded despite the incorrect message. * The SIP_OUTGOING flag was not set correctly for the dialog and I had to move some CC subscribe handling code as a result. * Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes in and the CC instance is not found, the 404 response was duplicated. JIRA AST-568 JIRA SWP-3493 ........ * apps/app_dial.c, main/channel.c, main/manager.c, /, apps/app_meetme.c, apps/app_fax.c, main/features.c, CHANGES, apps/app_queue.c, UPGRADE-1.8.txt: Merged revisions 320823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ * main/channel.c, /, main/format_cap.c, main/features.c, include/asterisk/channel.h: Merged revisions 320796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines Give zombies a safe channel driver to use. Recent crashes from zombie channels suggests that they need a safe home to goto. When a masquerade happens, the physical part of the zombie channel is hungup. The hangup normally sets the channel private pointer to NULL. If someone then blindly does a callback to the channel driver, a crash is likely because the private pointer is NULL. The masquerade now sets the channel technology of zombie channels to the kill channel driver. Related to the following issues: (issue #19116) (issue #19310) Review: https://reviewboard.asterisk.org/r/1224/ ........ 2011-05-25 15:43 +0000 [r320772] Gregory Nietsky * funcs/func_channel.c, CHANGES: CHANNEL(pickupgroup) Allow Setting / Reading the pickupgroup of a channel with func_channel.c (closes issue #19045) Reported by: irroot Review: https://reviewboard.asterisk.org/r/1148/ 2011-05-25 00:52 +0000 [r320717] Terry Wilson * /, addons/chan_mobile.c: Merged revisions 320716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24 May 2011) | 4 lines Cast data as char * before using S_OR This is required for compiling successfully under dev mode ........ 2011-05-23 18:00 +0000 [r320651] Richard Mudgett * main/manager.c, /, CHANGES: Merged revisions 320650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines Add ConnectedLineNum/Name headers to output of AMI action Status. * Add ConnectedLineNum and ConnectedLineName headers to the output of the AMI action Status. This makes it easier to find out who the channel is connected to without having to lookup BridgedChannel or when they are connected to an application (e.g.: VoiceMail) which has no bridged channel. * Bridged channels with no CallerID had "" instead of "" output, that might be a bug as "" was what older versions used. (closes issue #18158) Reported by: gareth Patches: svn-292308.diff uploaded by gareth (license 208) ........ 2011-05-23 16:28 +0000 [r320606] David Vossel * main/tcptls.c, /: Merged revisions 320568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320568 | dvossel | 2011-05-23 11:18:33 -0500 (Mon, 23 May 2011) | 14 lines Merged revisions 320562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011) | 9 lines Adds missing part to the ast_tcptls_server_start fails second attempt to bind patch. (closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls_2.patch uploaded by wdoekes (license 717) ........ ................ 2011-05-23 16:20 +0000 [r320579] Tilghman Lesher * /, configure, configure.ac: Merged revisions 320573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320573 | tilghman | 2011-05-23 11:19:32 -0500 (Mon, 23 May 2011) | 7 lines GNU libiconv uses symbol "libiconv_open" instead of "iconv_open". (closes issue #19344) Reported by: rohanl Patches: iconv-check.patch uploaded by rohanl (license 1284) ........ 2011-05-23 15:48 +0000 [r320561] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 320560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May 2011) | 4 lines Don't generate spurious "No: command not found" messages when running the configure script on a system that has neither gmime-config nor pkg-config. ........ 2011-05-23 14:40 +0000 [r320505] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 320504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | 10 lines Fixes segfault occuring in chan_sip.c at __set_address_from_contact Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve which is where the segfault was occuring due to null str. (closes issue #18857) Reported by: sybasesql Review: https://reviewboard.asterisk.org/r/1225/ ........ 2011-05-22 23:36 +0000 [r320446] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 320445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320445 | tilghman | 2011-05-22 18:34:57 -0500 (Sun, 22 May 2011) | 15 lines Merged revisions 320444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011) | 8 lines Don't crash when the connection fails. (closes issue #19250) Reported by: seadweller Patches: 20110514__issue19250.diff.txt uploaded by tilghman (license 14) Tested by: seadweller, sum ........ ................ 2011-05-20 21:40 +0000 [r320340] David Vossel * main/tcptls.c, /: Merged revisions 320338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320338 | dvossel | 2011-05-20 16:39:36 -0500 (Fri, 20 May 2011) | 14 lines Merged revisions 320271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011) | 8 lines Fixes issue with ast_tcptls_server_start failing on second attempt to bind. (closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls.patch uploaded by wdoekes (license 717) ........ ................ 2011-05-20 20:53 +0000 [r320238] Richard Mudgett * /, apps/app_meetme.c: Merged revisions 320237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ ................ ................ 2011-05-20 18:49 +0000 [r320181] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 320180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines This commit modifies the way polling is done on TLS sockets. Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ 2011-05-20 18:29 +0000 [r320178] Jonathan Rose * /, apps/app_voicemail.c: Merged revisions 320162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines Fixes an imapfolder related crash imapfolders being set in the general section of voicemail would cause the inbox folder name to change. Since sound file names are made based on the names of the folders, this would cause the audio related to that folder name to change and if Asterisk attempted to play it, the channel would instantly hang up when the audio file couldn't be found. This patch searches for the name of the folder first to leave existing behavior in tact and if that fails, it uses the normal inbox name to get the sound file instead. (closes issue #16104) Reported by: blkline Review: https://reviewboard.asterisk.org/r/1215/ ........ 2011-05-20 17:04 +0000 [r320058-320060] Richard Mudgett * /, main/features.c: Merged revisions 320059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) | 1 line Misc comment cleanup in features.c. ........ * main/channel.c, /, main/features.c: Merged revisions 320057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) | 19 lines Crash while transferring a call during DTMF feature timeout. When a call is being attended transferred during the time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so tech data is not available), making ast_dtmf_stream() segfault when it tries to send the DTMF digit (at least with SIP channels). Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256) * Check for zombies when ast_channel_bridge() returns. * Guarantee that the fo parameter value is initialized in ast_channel_bridge() before any returns. (closes issue #19116) Reported by: Irontec Tested by: rmudgett ........ 2011-05-20 16:27 +0000 [r320040] Jonathan Rose * funcs/func_strings.c, CHANGES: Adds STRREPLACE function Adds a new STRREPLACe function to func_strings.c that allows users to search and replace against a variable in the dialplan. (closes issue #18023) Reported by: wdoekes Review: https://reviewboard.asterisk.org/r/1219/ 2011-05-20 16:20 +0000 [r319998-320013] Richard Mudgett * /, apps/app_directed_pickup.c, main/features.c: Merged revisions 320007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines Change some variable names to make pickup code easier to understand. ........ * /, apps/app_directed_pickup.c, main/features.c: Merged revisions 319997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines Crash when using directed pickup applications. The directed pickup applications can cause a crash if the pickup was successful because the dialplan keeps executing. This patch does the following: * Completes the channel masquerade on a successful pickup before the application returns. The channel is now guaranteed a zombie and must not continue executing the dialplan. * Changes the return value of the directed pickup applications to return zero if the pickup failed and nonzero(-1) if the pickup succeeded. * Made some code optimizations that no longer require re-checking the pickup channel to see if it is still available to pickup. (closes issue #19310) Reported by: remiq Patches: issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, remiq, rmudgett Review: https://reviewboard.asterisk.org/r/1221/ ........ 2011-05-20 13:42 +0000 [r319867-319939] Jonathan Rose * /, channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ * /, main/features.c: Merged revisions 319866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | 11 lines Fix Randomize option on Park() The randomize option was generally not working like it should have at all on Park(). This patch restores intended functionality. (closes issue #18862) Reported by: davidw Tested by: jrose Review: https://reviewboard.asterisk.org/r/1222/ ........ 2011-05-19 18:12 +0000 [r319813] Mark Murawki * cel/cel_odbc.c, /: Merged revisions 319812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | 9 lines In cel_odbc, an uninitialized RWLIST is attempted to be locked. Added INIT and DESTROY for the RWLIST odbc_tables (closes issue #19331) Reported by: kobaz Patches: odbc_cel.patch uploaded by kobaz (license 834) ........ 2011-05-19 16:52 +0000 [r319759] Richard Mudgett * /, main/ccss.c: Merged revisions 319758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines CCSS generic agent with POTS and ISDN phones fail caller busy call-back test. If the following is true after a CCSS activation: * The generic agent is for an analog phone or ISDN phone. (Caller party) * The called party becomes available. * The caller party is not available. When the caller party becomes available, the caller is not alerted to the called party being available. The generic agent still thinks the caller is busy. * Fixed the generic agent device state event subscription to look for all device states that are considered available. * Encapsulated the device state test for CCSS generic device available in cc_generic_is_device_available(). Made the generic agent and monitor use the new function instead of the manually coded inline equivalent. JIRA AST-559 JIRA SWP-3462 ........ 2011-05-18 23:18 +0000 [r319530-319661] Terry Wilson * /, channels/chan_sip.c: Merged revisions 319654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 319552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines Unbreak the storing of registrations for restart The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq ........ * apps/app_dial.c, /: Merged revisions 319529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines Merged revisions 319528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ ................ ................ 2011-05-17 22:04 +0000 [r319471] Richard Mudgett * /, channels/misdn/isdn_lib.c: Merged revisions 319469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines Merged revision 319468 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on dialed channels. The use of mISDN HDLC mode is prevented if the mISDN dial technology option 'h1' is used when config option astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC mode. Instead of setting the channel to HDLC mode it is set to transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the logging message is correct, but the if condition is not. Make check the nodsp and hdlc flags. JIRA ABE-2787 JIRA SWP-3437 .......... ................ 2011-05-17 21:59 +0000 [r319470] Damien Wedhorn * channels/chan_skinny.c: Remove extraneous line variables. The vars were either explicitly or implicitly not used. 2011-05-17 20:13 +0000 [r319427] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message. The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 2011-05-17 12:54 +0000 [r319366-319368] Leif Madsen * /, apps/app_voicemail.c: Merged revisions 319367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) | 10 lines Don't create [general] voicemail context when using users.conf Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas ........ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 319365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) | 6 lines Make Debian init script lsb compliant (closes issue #18896) Reported by: manwe Patches: debian_init_lsb.patch uploaded by manwe (license 1223) ........ 2011-05-16 21:39 +0000 [r319316] Damien Wedhorn * channels/chan_skinny.c: Fix up skinny hints. Probably haven't been working for a couple of years. May still need some more love, but they are now working, both as a hint device and monitoring a hint. Changes centre around the long ago change to remove the requirement for a device name in a skinny line, and changes to the transmit_* functions. 2011-05-16 21:08 +0000 [r319262] Jonathan Rose * main/dsp.c: Merged revisions 319261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths. ........ 2011-05-16 20:41 +0000 [r319260] Richard Mudgett * /, main/ccss.c: Merged revisions 319259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines Deadlock between generic CCSS agent and native ISDN CCSS. Deadlock can occur when the generic CCSS agent is deleting duplicate CC offers and the native ISDN CC driver is processing an incoming CC message. The cc_core_instances container lock cannot be held when an agent or monitor callback is invoked without the possibility of a deadlock. * Make kill_duplicate_offers() remove the reference in cc_core_instances outside of the container lock. JIRA AST-566 JIRA SWP-3469 ........ 2011-05-16 18:21 +0000 [r319212] Terry Wilson * /, channels/chan_sip.c: Merged revisions 319204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319204 | twilson | 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) | 11 lines Merged revisions 319202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines Unlink a peer from peers_by_ip when expiring a registration Review: https://reviewboard.asterisk.org/r/1218/ ........ ................ 2011-05-16 15:58 +0000 [r319146] David Vossel * /, channels/chan_sip.c: Merged revisions 319145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319145 | dvossel | 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) | 9 lines Merged revisions 319144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines Fixes issue with peer ref-counting during handle_request_subscribe. (closes issue #19293) Reported by: irroot ........ ................ 2011-05-16 15:54 +0000 [r319143] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 319142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) | 8 lines Make sure tcptls_session exists before dereferencing it. (closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw ........ 2011-05-16 14:56 +0000 [r319087] Gregory Nietsky * channels/chan_sip.c, res/res_fax.c, CHANGES, channels/sip/include/sip.h: When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 2011-05-16 14:38 +0000 [r319086] Paul Belanger * /, configure, include/asterisk/autoconfig.h.in, configure.ac, res/res_http_post.c: Merged revisions 319085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) | 10 lines Support gmime-2.4 (closes issue #18863) Reported by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir (license 46) Tested by: tzafrir Review: https://reviewboard.asterisk.org/r/1213/ ........ 2011-05-16 14:29 +0000 [r319084] David Vossel * /, formats/format_wav.c: Merged revisions 319083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) | 5 lines Fixes Big Endian build issue. (closes issue #19298) Reported by: tzafrir ........ 2011-05-15 23:17 +0000 [r319024] Damien Wedhorn * channels/chan_skinny.c: Add activatesub and dialandactivate sub. When called, activatesub first cleans up the active sub and then handles the sub passed. dialandactivatesub first sets sub->exten and then calls activatesub. Revise handle_offhook to utilise the callid sent to chan_skinny. Some other minor fixes especially around d->hookstate (which still needs some more work). 2011-05-13 18:10 +0000 [r318918-318922] Brett Bryant * main/channel.c, /: Merged revisions 318921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) | 8 lines Fixes a segmentation fault in dynamic hints when a channel technology isn't loaded for a hint. (closes issue #18495) Reported by: bertrand Tested by: bertrand ........ * /, res/res_srtp.c: Merged revisions 318919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too much time has passed between sending audio. (closes issue #18206) Reported by: bernhardsi Patches: res_srtp_unhold.patch uploaded by bernhards (license 1138) Tested by: bernhards, notthematrix ........ * /, channels/chan_sip.c: Merged revisions 318917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) | 11 lines This patch allows TCP peers into the ast_db where they were previously restricted. (closes issue #18882) Reported by: cmaj Patches: patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt uploaded by cmaj (license 830) Tested by: cmaj ........ 2011-05-13 16:30 +0000 [r318869] Richard Mudgett * /, main/features.c: Merged revisions 318868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines CDR's are being written immediately on caller hangup. CDR's are being written immediately on caller hangup. The dialplan is not able to modify it in the h exten. The h exten in the initial context is not run before closing CDR's when the bridge is unlinked if a macro is active and does not have an h exten. * Make ast_bridge_call() check for an h exten in the current context and if a macro is active then the initial context. The first h exten found is then run before closing the CDR. (closes issue #18212) Reported by: leearcher Patches: issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ ........ 2011-05-13 08:33 +0000 [r318833] Damien Wedhorn * channels/chan_skinny.c: Move exten used for dialing from device to subchannel. There were some issues where if a simple switch was cancelled and a new switch started before the first had timed out where the d->exten would be used for both subchannels. This was bad leading to possible invalid extensions if some digits had been entered in the abandoned simple switch and the second one was completed before the first timed out, or the second would be cancelled because d->exten would be set to nothing on the time out of the first. 2011-05-13 01:55 +0000 [r318785] Matthew Nicholson * /, channels/sip/reqresp_parser.c: Merged revisions 318720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) | 4 lines Handle ipv6 addresses in the sent-by Via: field. This change fixes a regression in via header parsing and ipv6 handling. (closes issue #18951) ........ 2011-05-13 01:50 +0000 [r318784] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 318783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky ........ 2011-05-12 22:56 +0000 [r318672] Alec L Davis * /, channels/chan_sip.c, apps/app_directed_pickup.c, main/features.c, include/asterisk/features.h: Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ 2011-05-12 20:44 +0000 [r318600-318635] Damien Wedhorn * channels/chan_skinny.c: Consolidate setsubstate_* into setsubstate and use a switch. Consolidate the functions and add some debugging info. Allows to be able to set a substate without explicitly knowing what the state is. * channels/chan_skinny.c: Add setsubstate_onhook. Add the setsubstate_onhook to complete the initial substate handling procedures. Added dumpsub(sub, forcehangup) which is the common way of calling setsubstate_onhook. Dumpsub attempts to activate another sub after setting the current one onhook. 2011-05-11 18:52 +0000 [r318551-318552] Terry Wilson * /, channels/chan_sip.c: Merged revisions 318550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) | 2 lines Comment out the REF_DEBUG that slipped in during debugging ........ * /, channels/chan_sip.c: Merged revisions 318549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines Merged revisions 318548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines Clean up several chan_sip reference leaks Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ Review: https://reviewboard.asterisk.org/r/1210/ ........ ................ 2011-05-10 23:42 +0000 [r318500] Richard Mudgett * /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 318499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett ........ 2011-05-10 15:16 +0000 [r318437] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 318436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) | 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read(). ........ 2011-05-10 00:22 +0000 [r318400] Terry Wilson * /, channels/chan_sip.c: Merged revisions 318337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines Merged revisions 318331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines Don't offer video to directmedia callee unless caller offered it as well Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) ........ ................ 2011-05-09 23:16 +0000 [r318283-318352] Richard Mudgett * /, res/Makefile, res/res_features.exports.in (removed): Merged revisions 318351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines Remove references to res_features and its export file. The contents of res/res_features.c was moved to into main/features.c awhile ago. There is no longer any need for the res/Makefile to reference res_features or the res_features linker exports file to exist. ........ * /, main/features.c: Merged revisions 318282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines Hangup extension executed twice. When a user hangs up a call, in certain circumstances, the hangup extension can end up being executed twice: 1) If a call is bridged and the 'h' extension executes the Hangup application, then the 'h' extension will be executed twice. 2) If a call is bridged within a macro (Dial or Queue), it has its own 'h' extension, the main context also has an 'h' extension, and the macro 'h' extension executes the Hangup application, then both 'h' extensions will be executed. * Revert originally commited fix for #16106 and just set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The bridge code just executed an 'h' extension so the main PBX loop does not need to execute one as well. (issue #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd ........ 2011-05-09 17:13 +0000 [r318234] David Vossel * /, channels/chan_sip.c: Merged revisions 318233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318233 | dvossel | 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) | 14 lines Merged revisions 318230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) | 7 lines Fixes cases where sip_set_rtp_peer can return too early during media path reset. (closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) ........ ................ 2011-05-09 17:00 +0000 [r318232] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 318231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache ................ 2011-05-09 14:41 +0000 [r318194] Leif Madsen * main/app.c: Increase prepend filename length. (closes issue #19238) Reported by: byronclark Patches: increase_prepend_filename_length.patch uploaded by byronclark (license 1200) 2011-05-09 14:37 +0000 [r318162-318193] Jonathan Rose * main/features.c: Minor change to 318141 to improve parsing behavior. * /, configs/features.conf.sample: Merged revisions 318148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. ........ 2011-05-09 14:11 +0000 [r318143] Matthew Nicholson * main/channel.c, /: Merged revisions 318142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) | 9 lines Make indicate/control frames WRITE events on framehooks. Also, if a framehook returns a non-control frame, don't forward it to the channel. (closes issue #19251) Reported by: irroot Patches: (modified) framehook_indicate.patch2 uploaded by irroot (license 52) Tested by: irroot ........ 2011-05-09 13:56 +0000 [r318141] Jonathan Rose * main/features.c, CHANGES: Allows ParkedCall application to specify a parkinglot. When invoking the app parkedcall, the argument can now include '@parkinglot' after the extension. (closes issue #18777) Reported by: cartama Patches: 0018777.diff uploaded by cartama (license 1157) Review: https://reviewboard.asterisk.org/r/1209/ 2011-05-09 07:40 +0000 [r318106] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_callwait. If a call is made to a line that already has a call and the device is offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN. 2011-05-07 23:36 +0000 [r318056-318058] Russell Bryant * res/res_config_curl.c, /: Merged revisions 318057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 May 2011) | 8 lines res_config_curl: fix a crash with static realtime. (closes issue #18413) Reported by: jmls Patches: 20101202__issue18413.diff.txt uploaded by tilghman (license 14) Tested by: jmls ........ * /, channels/chan_iax2.c: Merged revisions 318055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) | 7 lines chan_iax2: Don't overwrite port found with an SRV lookup. (closes issue #17291) Reported by: jcovert Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551) ........ 2011-05-06 23:07 +0000 [r317996-318019] Damien Wedhorn * channels/chan_skinny.c: Only allow voicemail if substate is OFFHOOK or no channel active (UNSET). (closes issue #17901) Reported by: salecha * channels/chan_skinny.c: Rename sub->parent to sub->line. Improve readability of code, eg, (sub->parent == d->activeline) becomes (sub->line == d->activeline). * channels/chan_skinny.c: Move the hookstate from line to device. Long time coming, finally moving the hookstate from line to device. This may fix some issues where a device has multiple lines. Previously we had to run through all lines on a device to see if it was actually onhook or not. 2011-05-06 21:49 +0000 [r317968-317970] Russell Bryant * /, apps/app_meetme.c: Merged revisions 317969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines Use the right variable to print the time in a debug message. The original patch also increased some buffer sizes, but that was already done in this version. (closes issue #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded by sysreq (license 1009) ........ * /, apps/app_meetme.c: Merged revisions 317967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines Fix some more "set but unused" compiler warnings. ........ 2011-05-06 21:10 +0000 [r317920] David Vossel * res/res_rtp_asterisk.c, /: Merged revisions 317918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 May 2011) | 7 lines Fixes missing colon from To/From headers in RTCP manager events. (closes issue #18221) Reported by: clegall_proformatique Patches: 18221_1.patch uploaded by ebroad (license 878) ........ 2011-05-06 21:07 +0000 [r317843-317919] Russell Bryant * main/pbx.c, /: Merged revisions 317917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) | 7 lines Fix calculation of free RAM to make minmemfree option work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff uploaded by loic (license 1020) ........ * contrib/scripts/import-cdr-csv-mysql.pl (added): Add a cdr_csv to MySQL import script to contrib/scripts. (closes issue #17036) Reported by: precisenetworks Patches: import-cdr-csv-mysql.pl uploaded by precisenetworks (license 1010) * apps/app_userevent.c, CHANGES: Add the Uniqueid header to Userevent. (closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) * /, channels/chan_sip.c: Merged revisions 317867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. Don't duplicate variables on the sip_pvt. Just reset the variable list each time. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ * /, channels/chan_sip.c: Merged revisions 317865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't block doing silly deadlock avoidance. Just return and try again later. The funciton gets called often enough that it's fine. Also, this change was already made in trunk. (closes issue #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch uploaded by irroot (license 52) ........ * addons/app_mysql.c, /: Merged revisions 317837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines Fix a crash in the MySQL() application. This code was not handling channel datastores safely. The channel must be locked. (closes issue #17964) Reported by: wuwu Patches: issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71) Tested by: wuwu ........ 2011-05-06 19:23 +0000 [r317818-317833] Matthew Nicholson * CHANGES: Updated CHANGES to note the autoservice changes for pbx_lua * configs/extensions.lua.sample: Updated the sample pbx_lua config file to reflect autoservice changes. 2011-05-06 19:15 +0000 [r317807] Russell Bryant * /, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) | 7 lines Add a new sipfriends.sql for MySQL that has more fields in it. (closes issue #16399) Reported by: pabelanger Patches: sipfriends.sql.v3 uploaded by pabelanger (license 224) ........ 2011-05-06 19:14 +0000 [r317721-317806] Matthew Nicholson * pbx/pbx_lua.c, UPGRADE.txt: Default to starting an autoservice in pbx_lua. The autoservice is automatically stopped when applications are executed, so this shouldn't cause any problems. * pbx/pbx_lua.c, UPGRADE.txt: Make pbx_lua handle managing the autoservice better. Make autoservice_start() and autoservice_stop() return nothing. Also check if the autoservice flag is set before starting or stopping the autoservice and stop and start the autoservice when returning control to and getting control from the pbx engine. * UPGRADE.txt: Added note about changes in pbx_lua's behavior when applications do dialplan jumps * CHANGES: Use two spaces after periods for the recent pbx_lua change descriptions * CHANGES: Updated CHANGES for hints support in pbx_lua * pbx/pbx_lua.c, CHANGES: Detect Goto in pbx_lua. This code will actually detect any dialplan jump from any application that calls ast_explicit_goto(). This change is only being done in trunk as it may change the way some dialplans execute. 2011-05-06 16:23 +0000 [r317671] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 317670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines Fix SIP connected line updates. This patch fixes a couple SIP connected line update problems: 1) The connected line needs to be updated when the initial INVITE is sent if there is a peer callerid configured. Previously, the connected line information did not get reported until the call was connected so SIP could not report connected line information in ringing or progress messages. 2) The connected line should not be updated on initial connect if there is no connected line information. Previously, all it did was wipe out any default preset CONNECTEDLINE information set by the dialplan with empty strings. (closes issue #18367) Reported by: GeorgeKonopacki Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ ........ 2011-05-06 08:21 +0000 [r317596] Terry Wilson * /, apps/app_queue.c: Merged revisions 317584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines Merged revisions 317575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ ................ ................ 2011-05-05 23:47 +0000 [r317426-317531] Russell Bryant * Makefile, /: Merged revisions 317530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) | 10 lines If the configure script runs, force a rebuild of menuselect-tree. Some contents in the menuselect tree are dependent on configure script parameters, namely --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis Patches: issue_17219.rev1.txt uploaded by russell (license 2) ........ * /, contrib/realtime/mysql/queue_log.sql, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) | 9 lines Fix some more realtime MySQL schema issues. (closes issue #18537) Reported by: denzs Patches: sipfriends.sql.svndiff uploaded by denzs (license 1182) queue_log.sql.svndiff uploaded by denzs (license 1182) meetme.sql.svndiff uploaded by denzs (license 1182) ........ * /, contrib/realtime/mysql/meetme.sql, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) | 8 lines Fix some errors in sample MySQL realtime schema files. (closes issue #18915) Reported by: Dovid Patches: sipfriends.patch uploaded by Dovid (license 652) meetme.patch uploaded by Dovid (license 652) ........ * CHANGES, res/res_calendar.c: Add "calendar show types" CLI command. (closes issue #18246) Reported by: junky Patches: calendar_types.diff uploaded by junky (license 177) * cel/cel_pgsql.c, UPGRADE.txt, configs/cel_pgsql.conf.sample, CHANGES: Add CEL extra field to cel_pgsql. (closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) * /, cdr/cdr_syslog.c: Merged revisions 317480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011) | 8 lines Don't lose cdr_syslog config on a reload. (closes issue #18679) Reported by: enegaard Patches: issue18679_seanbright.patch uploaded by seanbright (license 71) Tested by: enegaard ........ * channels/chan_unistim.c, channels/chan_usbradio.c, channels/chan_dahdi.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_console.c, channels/chan_oss.c, channels/chan_mgcp.c, channels/misdn_config.c: Merged revisions 317478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ * /, pbx/pbx_lua.c: Merged revisions 317476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011) | 8 lines Add a datastore fixup to fix a pbx_lua crash. (closes issue #19055) Reported by: jamhed Patches: lua_datastore_fixup1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, jamhed ........ * cel/cel_pgsql.c, channels/chan_jingle.c, channels/sip/sdp_crypto.c, res/res_config_odbc.c, /, channels/chan_sip.c, res/res_crypto.c, pbx/pbx_lua.c, channels/iax2-provision.c, pbx/pbx_dundi.c, channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c: Merged revisions 317474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) | 2 lines Fix more "set but unused" warnings. ........ * /, main/dsp.c: Merged revisions 317429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) | 5 lines Only display inband DTMF warning if inband DTMF detection is enabled. (closes issue #18901) Reported by: irroot ........ * /, apps/app_rpt.c: Merged revisions 317427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) | 7 lines Fix potential memory leak, and use of uninitialized memory. (closes issue #16476) Reported by: junky Patches: M16476.diff uploaded by junky (license 177) ........ * main/manager.c, /: Merged revisions 317425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317425 | russell | 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) | 7 lines Add missing ActioID handling to Events action. (closes issue #18949) Reported by: edersohe Patches: 0018949.patch uploaded by edersohe (license 1228) ........ 2011-05-05 21:20 +0000 [r317395] Sean Bright * main/asterisk.c: Add some new editline bindings by default, and allow for user specified configuration. I excluded the part of this patch that used the HOME environment variable since the built-in editline library goes to great lengths to disallow that. Instead only settings the EDITRC environment variable will use a user specified file. Also, the default environment variable use to determine the edit more is AST_EDITMODE instead of AST_EDITOR (although the latter is still supported). (closes issue #15929) Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) Tested by: seanbright 2011-05-05 20:46 +0000 [r317382] Damien Wedhorn * channels/chan_skinny.c: Move hold stuff to the setsubstate arrangement. skinny_hold moved to setsubstate_hold and skinny_unhold integrated into setsubstate_connected. Removed sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on answering a SUBSTATE_RINGIN sub (was a typo). 2011-05-05 20:27 +0000 [r317377] Sean Bright * /, addons/res_config_mysql.c: Merged revisions 317370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines Don't duplicate our data on the stack and just use the MYSQL_ROW directly. With large result sets we were blowing out the stack. (closes issue #19090) Reported by: mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by seanbright (license 71) Tested by: mickecarlsson ........ 2011-05-05 19:56 +0000 [r317337] Russell Bryant * /, apps/app_queue.c: Merged revisions 317336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) | 7 lines Increase buffer size to be PATH_MAX for a path. (closes issue #19239) Reported by: byronclark Patches: queue_announce_length.patch uploaded by byronclark (license 1200) ........ 2011-05-05 19:33 +0000 [r317334] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 317283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines Resolves a deadlock that occurs during sip_new This is based on an uncommitted patch by jpeeler for the issue. Instead of relocking and then unlocking the channel though, we keep the lock on the channel until we are finished doing what we need to the channel. (closes issue #18441) Reported by: Alric ........ 2011-05-05 18:46 +0000 [r317282] Russell Bryant * /, channels/chan_sip.c: Merged revisions 317281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317281 | russell | 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) | 29 lines Merged revisions 317255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines Merged revisions 317211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) ........ ................ ................ 2011-05-05 18:09 +0000 [r317198] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 317196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) | 8 lines Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer abruptly disappears. This mostly occurs after a successful registration. (closes issue #17544) Reported by: marcelloceschia Patches: (modified) tcptls.patch uploaded by st (license 907) ........ 2011-05-05 18:08 +0000 [r317197] David Vossel * bridges/bridge_softmix.c, funcs/func_jitterbuffer.c: Fixes reliability issues with func_jitterbuffer's usage in the new ConfBridge application. 2011-05-05 15:06 +0000 [r317059-317105] Leif Madsen * /, contrib/scripts/safe_asterisk: Merged revisions 317104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317104 | lmadsen | 2011-05-05 11:04:24 -0400 (Thu, 05 May 2011) | 15 lines Merged revisions 317102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) | 8 lines Disable console colourization inside safe_asterisk checks. (closes issue #19213) Reported by: lefoyer Patches: issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by wdoekes (license 717) Tested by: wdoekes, lefoyer ........ ................ * Makefile, configs/cel.conf.sample, /: Merged revisions 317058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) | 7 lines Remove unused directory and clear up some documentation. (closes issue #19193) Reported by: bchia Patches: cel-csv.diff uploaded by lathama (license 1028) Tested by: lathama, Marquis42 ........ 2011-05-05 09:03 +0000 [r316994-317026] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_congestion and setsubstate_progress. Move handling of both state handling from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants for congestion. Added various states to substate2str and added these states where applicable for other set_substate_ procs. * channels/chan_skinny.c: Add setsubstate_busy. Move handling of setting busy state from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants (eg busy(10); hangup() in the dialplan now gives a busy indication for 10 secs and then hangs up. 2011-05-05 07:09 +0000 [r316962] Stefan Schmidt * main/astobj2.c: Adding the Move to Front Hash functionality Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists. (closes issue #19233) Reported by: schmidts Review: https://reviewboard.asterisk.org/r/1201/ 2011-05-05 02:34 +0000 [r316920] Sean Bright * main/manager.c, /, main/http.c, main/utils.c: Merged revisions 316917-316919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) | 5 lines Make sure that tcptls_session is properly initialized. (issue #18598) Reported by: ksn ........ r316918 | seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 lines Look at the correct buffer for our digest info instead of an empty one. (issue #18598) Reported by: ksn ........ r316919 | seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 lines Use the correct HTTP method when generating our digest, otherwise we always fail. When calculating the 'A2' portion of our digest for verification, we need the HTTP method that is currently in use. Unfortunately our mapping function was incorrect, resulting in invalid hashes being generated and, in turn, failures in authentication. (closes issue #18598) Reported by: ksn ........ 2011-05-04 21:44 +0000 [r316885] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_ringout (equivalent to AST_STATE ringing). Renamed previous setsubstate_ringout to setsubstate_dialing for a state when attempting to dial a number, substate ringout now for when core has indicated that the channel is actually ringing on the other end. Also added substate2str for debugging purposes. 2011-05-04 18:57 +0000 [r316832] Richard Mudgett * /, apps/app_meetme.c: Merged revisions 316831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines Wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested by: rmudgett ........ 2011-05-04 16:42 +0000 [r316798] David Vossel * channels/chan_sip.c, CHANGES: Reverts rev 316218 as it breaks parsing the [general] section of sip.conf. The functionality this patch attempts to achieve should already be possible using [general](+) in the config file. issue #17957 2011-05-04 16:17 +0000 [r316664-316711] Sean Bright * /, apps/app_voicemail.c: Merged revisions 316709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316709 | seanbright | 2011-05-04 12:15:32 -0400 (Wed, 04 May 2011) | 22 lines Merged revisions 316708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400 (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) | 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ ................ * main/manager.c, /: Merged revisions 316663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316663 | seanbright | 2011-05-04 10:35:05 -0400 (Wed, 04 May 2011) | 8 lines Only return a single error via AMI when requesting a forbidden action. (closes issue #19216) Reported by: oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ 2011-05-04 14:26 +0000 [r316618-316657] David Vossel * /, apps/app_chanspy.c: Merged revisions 316650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316650 | dvossel | 2011-05-04 09:25:03 -0500 (Wed, 04 May 2011) | 15 lines Merged revisions 316644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) | 9 lines Fixes one-way-audio when chanspy activated with the 'o' option (closes issue #18382) Reported by: jkister Patches: 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel ........ ................ * /, channels/chan_sip.c: Merged revisions 316617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316617 | dvossel | 2011-05-04 08:44:41 -0500 (Wed, 04 May 2011) | 19 lines Merged revisions 316616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines Fixes session-timers=refuse not being enforced for *caller* During handle_request_invite, the session timer mode was retrieved from a cached variable. This patch forces a peer lookup of the session timer mode in the case of an incoming invite. (closes issue #18804) Reported by: wdoekes Patches: issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel (license 671) ........ ................ 2011-05-04 08:25 +0000 [r316552-316584] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_ringin. Added setsubstate_ringin. skinny_call now calls sss_ringin rather than inline. Fixed previous issue so that setsubstate_connected now use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should be queued. * channels/chan_skinny.c: Make skinny_answer use setsubsate_connected. Cosolidated the code so that skinny_answer now uses the setsubstate procedures rather than doing the handling inline. 2011-05-04 07:13 +0000 [r316520] Tzafrir Cohen * autoconf/ast_check_pwlib.m4, /, configure: Merged revisions 316193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316193 | tzafrir | 2011-05-03 13:57:16 +0300 (ג', 03 מאי 2011) | 8 lines Re-fix bashism in ./configure: s/let/$(( ))/ A forward-port in r278985 accidentally re-introduced issue 17485. Fixing it. Thanks to Jilles Tjoelker for the good report. (closes issue #17485) ........ 2011-05-04 07:10 +0000 [r316519] Damien Wedhorn * channels/chan_skinny.c: Cleanup skinny callinfo. Cosolidated the working out of the callinfo to be sent into transmit_callinfo. Replaced ambiguous sub->outgoing with calldirection which can be SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny protocol). 2011-05-04 02:39 +0000 [r316477] Sean Bright * /, apps/app_meetme.c: Merged revisions 316476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316476 | seanbright | 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) | 17 lines Merged revisions 316475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines Honor the C option to MeetMe when L is passed. This fixes a case that r304773 and friends missed. (closes issue #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff uploaded by var (license 1227) Tested by: seanbright ........ ................ 2011-05-04 00:13 +0000 [r316428-316430] Tilghman Lesher * /, addons/cdr_mysql.c, addons/res_config_mysql.c: Merged revisions 316429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316429 | tilghman | 2011-05-03 19:12:25 -0500 (Tue, 03 May 2011) | 7 lines Escape column names in case they contain illegal characters ('-') or reserved words. (closes issue #19063) Reported by: festr Patches: patch uploaded by festr (license 443) ........ * channels/chan_sip.c, CHANGES: If multiple [general] contexts occur from sip.conf (usually due to external includes), merge them. The original implementation of this did the merging of all contexts with the same name in the realtime layer, but that implementation severely breaks drivers which use the same context name (e.g. iax.conf, type={peer,user}). Therefore, the implementation needs to do the merging for particular entries only, based upon what contexts would allow that in the channel driver itself. This implementation is for chan_sip only, but others could be added in the future. (closes issue #17957) Reported by: marcelloceschia Patches: chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079) Tested by: tilghman 2011-05-03 22:16 +0000 [r316337] Russell Bryant * /, channels/chan_skinny.c, pbx/pbx_dundi.c, channels/chan_mgcp.c: Merged revisions 316336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) | 8 lines Use htons() instead of ntohs() in some places. (closes issue #19200) Reported by: wdoekes Patches: issue19200-trunk.patch uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........ 2011-05-03 22:07 +0000 [r316335] David Vossel * main/channel.c, /: Merged revisions 316334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) | 8 lines Fixes framehook segfault on indicate (closes issue #19215) Reported by: irroot Patches: framehook_indicate.patch uploaded by irroot (license 52) ........ 2011-05-03 21:48 +0000 [r316333] Russell Bryant * /, apps/app_minivm.c: Merged revisions 316331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316331 | russell | 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011) | 2 lines Resolve another warning. ........ 2011-05-03 21:45 +0000 [r316332] David Vossel * channels/chan_local.c, /: Merged revisions 316330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316330 | dvossel | 2011-05-03 16:37:59 -0500 (Tue, 03 May 2011) | 24 lines Merged revisions 316329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500 (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ ........ ................ ................ 2011-05-03 20:45 +0000 [r316293] Russell Bryant * channels/chan_unistim.c, main/udptl.c, main/fskmodem_float.c, main/rtp_engine.c, /, res/res_musiconhold.c, apps/app_ices.c, apps/app_followme.c, main/config.c, main/channel.c, main/cdr.c, channels/chan_phone.c, funcs/func_enum.c, main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, main/features.c, main/plc.c, res/res_agi.c, apps/app_amd.c, main/pbx.c, res/res_fax.c, formats/format_wav.c, apps/app_festival.c, channels/chan_agent.c, apps/app_originate.c, apps/app_queue.c, codecs/lpc10/dyptrk.c, include/asterisk/linkedlists.h, main/file.c, main/audiohook.c, pbx/pbx_config.c, main/asterisk.c, main/dsp.c, res/res_calendar.c, apps/app_voicemail.c: Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ 2011-05-03 19:22 +0000 [r316240] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_pri.c: Merged revisions 316224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) | 16 lines The dahdi_hangup() call does not clean up the channel fully. After dahdi_hangup() has supposedly hungup an ISDN channel there is still traffic on the S0-bus because the channel was not cleaned up fully. Shuffled the hangup code to include some missing cleanup. Also fixed some code formatting in the area. I think the primary missing clean up code was the call to tone_zone_play_tone() to turn off any active tones on the channel. (closes issue #19188) Reported by: jg1234 Patches: issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested by: jg1234 ........ 2011-05-03 19:00 +0000 [r316216-316218] David Vossel * /, channels/chan_sip.c: Merged revisions 316217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) | 9 lines Never put the Require: timer header in an Invite. This has already been discussed and should have been resolved earlier. View revsion 285565's log for more information about why it is important to not put timer in the Require header. (closes issue #18704) Reported by: mfrager ........ * /, res/res_odbc.c: Merged revisions 316215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316215 | dvossel | 2011-05-03 13:49:48 -0500 (Tue, 03 May 2011) | 9 lines Fixes a random crash (NULL reference) in res_odbc.c. (closes issue #19180) Reported by: pruiz Patches: tmp.diff uploaded by pruiz (license 1152) Tested by: pruiz, seanbright ........ 2011-05-03 18:23 +0000 [r316213] Sean Bright * main/manager.c, /: Merged revisions 316206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) | 8 lines If we aren't interested in events, don't generate the FullyBooted event on AMI login. (closes issue #19089) Reported by: bklang Patches: issue19089-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ 2011-05-02 19:15 +0000 [r316095] Tilghman Lesher * funcs/func_curl.c, /: Merged revisions 316094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316094 | tilghman | 2011-05-02 14:09:55 -0500 (Mon, 02 May 2011) | 15 lines Merged revisions 316093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) | 8 lines More possible crashes based upon invalid inputs. (closes issue #18161) Reported by: wdoekes Patches: 20110301__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes ........ ................ 2011-05-02 15:58 +0000 [r316054] Paul Belanger * apps/app_meetme.c: Formatting change, remove red blobs 2011-04-27 19:15 +0000 [r315895] Matthew Nicholson * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged revisions 315894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines Merged revisions 315893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) ........ ................ ................ 2011-04-27 17:51 +0000 [r315855-315856] David Vossel * apps/app_confbridge.c: Makes the new ConfBridge join and leave sounds be used by default rather than beep and beeperr. * main/channel.c: Clears exception flag during ast_read when func_jitterbuffer is enabled 2011-04-27 15:56 +0000 [r315811] Russell Bryant * /, main/asterisk.c: Merged revisions 315810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) | 2 lines Set the copyright year to 2011 in the startup message. ........ 2011-04-27 12:37 +0000 [r315766] Leif Madsen * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 315765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011) | 4 lines Enable Russian core sound selection in menuselect. (closes issue #18724) Reported by: pbxware ........ 2011-04-26 23:10 +0000 [r315670-315675] Terry Wilson * /, channels/chan_sip.c: Merged revisions 315673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315673 | twilson | 2011-04-26 15:56:19 -0700 (Tue, 26 Apr 2011) | 25 lines Merged revisions 315672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson ........ ................ ................ * channels/chan_sip.c: Make sure to create the caps structure for autocreated peers Because crashing is bad. * apps/app_dial.c, main/features.c, apps/app_queue.c: Merged revisions 315644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ ................ ................ 2011-04-26 22:18 +0000 [r315649] Richard Mudgett * main/pbx.c, /: Merged revisions 315645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines The 'e' special extension fails to trigger in at least two cases. The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if any of them do not exist. Many of the places the 'e' extension was supposed to be invoked fail because the priority was set wrong. There were two places where the 'e' extension was not even checked for fall back. * Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T' extension check and added the 'e' extension as a fall back to the two missing locations. * Prioritized and optimized some hangup tests associated with the 'e' extension. (closes issue #19136) Reported by: kshumard Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1196/ ........ 2011-04-26 19:38 +0000 [r315504] Tilghman Lesher * include/asterisk/select.h, /: Merged revisions 315503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315503 | tilghman | 2011-04-26 14:32:50 -0500 (Tue, 26 Apr 2011) | 28 lines Merged revisions 315502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ ................ ................ 2011-04-26 18:02 +0000 [r315453] Richard Mudgett * apps/app_dial.c, /: Merged revisions 315452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) | 1 line Add missing set of name valid flag when dialing. ........ 2011-04-26 17:41 +0000 [r315447] Russell Bryant * channels/chan_local.c, /: Merged revisions 315446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 Apr 2011) | 14 lines chan_local: resolve a deadlock. This patch resolves a fairly complex deadlock that can occur with the combination of chan_local and a dialplan switch, such as dynamic realtime extensions, which pulls autoservice into the picture when doing a dialplan lookup. (closes issue #18818) Reported by: nic Patches: issue18818.patch uploaded by jthurman (license 614) 18818.v1.txt uploaded by russell (license 2) Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik ........ 2011-04-26 02:21 +0000 [r315395] Paul Belanger * /, pbx/pbx_config.c: Merged revisions 315394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315394 | pabelanger | 2011-04-25 22:18:50 -0400 (Mon, 25 Apr 2011) | 14 lines Merged revisions 315393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr 2011) | 7 lines Add back CLI command 'dialplan save' (closes issue #19140) Reported by: lmadsen Patches: __20110419_dialplan_save.patch.txt uploaded by lmadsen (license 10) ........ ................ 2011-04-25 21:55 +0000 [r315350] Richard Mudgett * /, channels/chan_mgcp.c: Merged revisions 315349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 Apr 2011) | 9 lines When using MGCP realtime gateway definitions, random crashes occur. Fixed incorrect linked list node removal for realtime gateways. (closes issue #18291) Reported by: nahuelgreco Patches: dangling-pointers-when-pruning.patch uploaded by nahuelgreco (license 162) ........ 2011-04-25 19:40 +0000 [r315214-315260] Russell Bryant * /, formats/format_wav.c: Merged revisions 315259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315259 | russell | 2011-04-25 14:37:32 -0500 (Mon, 25 Apr 2011) | 24 lines Merged revisions 315258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315258 | russell | 2011-04-25 14:31:44 -0500 (Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) | 10 lines Be more flexible with unknown chunks in wav files. This patch makes format_wav ignore unknown chunks instead of erroring out on them. (closes issue #18306) Reported by: jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156) ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 315213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315213 | russell | 2011-04-25 14:04:28 -0500 (Mon, 25 Apr 2011) | 14 lines Merged revisions 315212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) | 7 lines Don't link non-cached realtime peers into the peers_by_ip container. (closes issue #18924) Reported by: wdoekes Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717) ........ ................ 2011-04-25 07:17 +0000 [r315054] Alec L Davis * channels/chan_local.c, /: Merged revisions 315053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315053 | alecdavis | 2011-04-25 19:14:32 +1200 (Mon, 25 Apr 2011) | 23 lines Merged revisions 315052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200 (Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) | 11 lines chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) ........ ................ ................ 2011-04-22 23:01 +0000 [r315002] Alec L Davis * channels/chan_dahdi.c, /: Merged revisions 315001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 Apr 2011) | 12 lines chan_dahdi: Can't return to normal ring after distinctive ring on FXS clear a previous distinctivering pattern before each new call (closes issue #18985) Reported by: bromont Patches: bug18985.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, bromont ........ 2011-04-22 21:33 +0000 [r314960] Matthew Nicholson * /, channels/chan_agent.c: Merged revisions 314959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314959 | mnicholson | 2011-04-22 16:20:08 -0500 (Fri, 22 Apr 2011) | 24 lines Merged revisions 314958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500 (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) | 4 lines Don't hold the pvt lock while streaming a file. ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 ........ ................ ................ 2011-04-22 14:49 +0000 [r314824] Tzafrir Cohen * channels/chan_unistim.c, /, res/res_fax_spandsp.c: Merged revisions 314779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | 2 lines Fix a few typos (shown by Lintian) ........ 2011-04-22 14:08 +0000 [r314781] Russell Bryant * /, res/res_agi.c: Merged revisions 314780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314780 | russell | 2011-04-22 09:02:23 -0500 (Fri, 22 Apr 2011) | 18 lines Merged revisions 314778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) | 11 lines Initialize buffers in getvar and getvarfull. Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz ........ ................ 2011-04-21 22:53 +0000 [r314733-314735] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: Implement AMI action PRIShowSpans. PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI spans. It is similar to the CLI command "pri show spans". (closes issue #15980) Reported by: dwery * channels/sig_pri.c: Simplify sig_pri.c:build_status(). * channels/chan_dahdi.c, /: Merged revisions 314732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 Apr 2011) | 1 line Correct DAHDIShowChannels XML documentation. ........ 2011-04-21 18:32 +0000 [r314666] Matthew Nicholson * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, main/http.c, configs/sip.conf.sample, configs/skinny.conf.sample, channels/sip/include/sip.h, configs/http.conf.sample: Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ 2011-04-21 18:11 +0000 [r314598] David Vossel * configs/confbridge.conf.sample (added), apps/confbridge (added), bridges/bridge_softmix.c, UPGRADE.txt, include/asterisk/channel.h, res/res_musiconhold.c, CHANGES, apps/confbridge/conf_config_parser.c (added), main/channel.c, include/asterisk/bridging_technology.h, bridges/bridge_builtin_features.c, apps/confbridge/include/confbridge.h (added), apps/Makefile, include/asterisk/bridging_features.h, include/asterisk/bridging.h, include/asterisk/dsp.h, apps/app_confbridge.c, apps/confbridge/include (added), main/bridging.c, main/dsp.c: New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ 2011-04-21 00:29 +0000 [r314551] Terry Wilson * /, channels/chan_sip.c: Merged revisions 314550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314550 | twilson | 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) | 13 lines Merged revisions 314549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines Don't allocate more space than necessary for a sip_pkt This extra allocation is a hold-over from when pkt->data was a character array. Now that it is an allocated string, just allocate enough for the sip_pkt. ........ ................ 2011-04-20 20:52 +0000 [r314509] David Vossel * main/channel.c, main/abstract_jb.c, funcs/func_jitterbuffer.c (added), include/asterisk/channel.h, CHANGES, include/asterisk/abstract_jb.h: Introduction of the JITTERBUFFER dialplan function. Review: https://reviewboard.asterisk.org/r/1157/ 2011-04-20 19:56 +0000 [r314471] Shaun Ruffell * codecs/codec_dahdi.c: codec_dahdi: DAHDI still advertises formats using the old bitfields. Previously, the DAHDI format bit fields matched up with the Asterisk bitfields. Since the Asterisk codec bit fields were replaced in r306010, codec_dahdi needs to contain the formats itself. In the future, the DAHDI formats should either change to something other than bitfields, or the bitfields need to move from include/dahdi/kernel.h to include/dahdi/user.h. Signed-off-by: Shaun Ruffell 2011-04-20 16:55 +0000 [r314418] Richard Mudgett * /, include/asterisk/frame.h: Merged revisions 314417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 Apr 2011) | 1 line AST_CONTROL_XXX comment changes. ........ 2011-04-20 16:37 +0000 [r314415] David Vossel * codecs/codec_resample.c: Fixes error with frame datalen being calculated from samples when this is not allwaya accurate. 2011-04-20 05:28 +0000 [r314359] Terry Wilson * main/lock.c, /: Merged revisions 314358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) | 4 lines Initialize track pointer ast_reentrancy_init checks to see if it is NULL before initializing with calloc ........ 2011-04-19 15:42 +0000 [r314204-314252] Leif Madsen * main/tcptls.c, /: Merged revisions 314251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) | 8 lines Use SSLv23_client_method instead of old SSLv2 only. (closes issue #19095) (closes issue #19138) Reported by: tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by: russell, chazzam ........ * /, funcs/func_channel.c: Merged revisions 314206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314206 | lmadsen | 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) | 14 lines Merged revisions 314205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) | 6 lines Remove duplicate documentation from func_channel.c (closes issue #18970) Reported by: IgorG Patches: func_channel.c.doc.diff uploaded by IgorG (license 20) ........ ................ * apps/app_dial.c, /: Merged revisions 314203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314203 | lmadsen | 2011-04-19 09:24:25 -0500 (Tue, 19 Apr 2011) | 15 lines Merged revisions 314202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) | 7 lines Update seconds to milliseconds in ast_verb output. (closes issue #19084) Reported by: smurfix Patches: app_dial.patch uploaded by smurfix (license 547) Tested by: lmadsen, smurfix ........ ................ 2011-04-19 08:22 +0000 [r314158] Olle Johansson * apps/app_meetme.c: Add explanation of strange flag setup in app_meetme (stolen from Mark's message to asterisk-dev) 2011-04-18 19:48 +0000 [r314079-314116] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Problems with ISDN MWI to phones. The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 * /, res/res_agi.c: Merged revisions 314069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines The AsyncAGI command loop is lax in the value it returns for the return status. * Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the command loop such as hangup would return SUCCESS. * The "asyncagi break" command now returns SUCCESS and is now the only way to break the command loop with that status. Previously, it returned FAILED. * The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event is not sent. Previously, this happened because of an error setting up the AGI pipes. * All executed AGI commands now get an AsyncAGI Exec result event. Previously, if the command returned failure (because of hangup), the command loop just exited with FAILURE and did not send the AsyncAGI Exec result event. * Makes sure that the channel frame queue is empty on hangup. Review: https://reviewboard.asterisk.org/r/1183/ ........ * apps/app_dial.c, /: Merged revisions 314068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines Unclear code in app_dial.c. Make code formatting clear. (closes issue #19134) Reported by: oej ........ 2011-04-18 16:22 +0000 [r314018-314078] David Vossel * /, channels/chan_sip.c: Merged revisions 314067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines Remove the need for deadlock avoidance in chan_sip do_monitor. Deadlock avoidance between the sip pvt and the pvt->owner is very difficult. Now that channel's are ao2 objects, this complication is no longer necessary. It turns out the pvt's msg queue only exists because of deadlock avoidance (when deadlock avoidance fails msgs were added to a queue to be processed later), so this goes away as well. The technique used in the new sip_lock_pvt_full() function should be used as a template for replacing all locations where deadlock avoidance occurs between a channel tech_pvt and the pvt's owner. My hope is that this will begin a reversal of the invalid channel driver locking architecture we have been using for so long. This patch also resolves an issue where the pvt->owner gets unlocked during processing the msg queue. (closes issue #18690) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........ * main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 314017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines sip codec negotiation of dynamic rtp payloads error fix This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ ........ 2011-04-17 09:28 +0000 [r313980] Damien Wedhorn * channels/chan_skinny.c: Consolidate all new call calls to run through new setsubstate_ringout. (closes issue #17907) Reported by: wedhorn Patches: cleanup.stateringout.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn 2011-04-17 01:28 +0000 [r313907-313944] Alexandr Anikin * addons/chan_ooh323.c: fix compile error from r313907 * addons/chan_ooh323.c: fix trivial error with set_max_datagram on pvt->udptl 2011-04-15 15:20 +0000 [r313867] Jonathan Rose * /, main/cli.c: Merged revisions 313860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313860 | jrose | 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) | 17 lines Merged revisions 313859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | 10 lines Fix a Tab Completion bug that occurs due to multiple matches on a substring. Makes word_match function in cli.c repeat a search for a command string until a proper match is found or the string is searched to the last point. (closes issue #17494) Reported by: ffossard Review: https://reviewboard.asterisk.org/r/1180/ ........ ................ 2011-04-14 21:53 +0000 [r313822] Terry Wilson * res/res_rtp_asterisk.c: Sets video mark bit on format field correctly This fixes a regression in the media architecture change where video frames did not have their video mark set correctly. dvossel wrote this. twilson kindly committed this, mmichelson found the bug. 2011-04-14 21:02 +0000 [r313606-313781] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 313780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines Leftover debug messages unconditionally sent to the console. Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config option enabled outputs the following debug messages unconditionally: Dialing T1847555121 on 1 Dialing www2w on 1 * Made debug messages in my_dial_digits() normal debug messages that do not get output unless enabled. * Reworded some debug messages in my_dial_digits() to be clearer. * Replace strncpy() with ast_copy_string() in my_dial_digits() which does the same job better. (closes issue #18847) Reported by: vmikhelson Tested by: rmudgett ........ * CREDITS, main/ccss.c, configs/ccss.conf.sample: Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => *823000,hint,ccss:sip/3000 You would then subscribe a BLF button to *823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ * /, res/res_agi.c: Merged revisions 313700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011) | 5 lines Revert flushing stale AsyncAGI commands from -r313615. It looks like it was intentional to leave any commands or in-flight commands in the queue in case Async AGI is run again on the call. ........ * /, res/res_agi.c: Merged revisions 313658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313658 | rmudgett | 2011-04-13 12:47:43 -0500 (Wed, 13 Apr 2011) | 2 lines Miscellaneous AGI diagnostic message cleanup and code optimization. ........ * /, res/res_agi.c: Merged revisions 313615 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011) | 5 lines * Add missing channel lock to handle_cli_agi_add_cmd(). * Flush any Async AGI commands left over from earlier Async AGI control of the call. ........ * main/channel.c, /, res/res_agi.c: Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ 2011-04-13 15:49 +0000 [r313528] Leif Madsen * configs/iax.conf.sample, configs/users.conf.sample, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h: Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ 2011-04-13 15:23 +0000 [r313527] Richard Mudgett * /, apps/app_dumpchan.c: Merged revisions 313517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines Bring the dumpchan application inline with "core show channel". * Added fields that are in "core show channel" to dumpchan output. * Fixed reuse of formatbuf before the previous string stored there was used by snprintf. All output strings now have their own buffer. * Adjusted the buffer sizes to not be so abusive of the stack now that there are more buffers. Change requested by oej. ........ 2011-04-12 21:59 +0000 [r313482] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooLogChan.h, addons/chan_ooh323.c, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/ooports.c, addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooh323ep.h, addons/ooh323c/src/ootypes.h, addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooSocket.h, addons/ooh323c/src/ooq931.c: IPv6 support for chan_ooh323 IPv6 support for ooh323, bindaddr, peers and users ip can be IPv4 or IPv6 addr correction for multi-homed mode (0.0.0.0 or :: bindaddr) can work in dual 6/4 mode with :: bindaddr gatekeeper mode isn't supported in v6 mode while (issue #18278) Reported by: may213 Patches: ipv6-ooh323.patch uploaded by may213 (license 454) Review: https://reviewboard.asterisk.org/r/1004/ 2011-04-12 18:53 +0000 [r313437-313438] Jonathan Rose * /: blocking fix from 313436 that was already made in this commit * channels/chan_dahdi.c, /: Merged revisions 313435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also went ahead and fixed the problem it introduces before committing. ........ r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line fixing stupid mistake with putting code before variable declaration ........ Merged revisions 313433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ ........ ........ 2011-04-11 23:20 +0000 [r313367-313383] Richard Mudgett * apps/app_dial.c, /: Merged revisions 313368-313369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines Backport a restructuring change from trunk to make the next change stand out. ........ r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines Frames from the inbound channel should go to all outbound channels in app_dial.c. In app_dial.c:wait_for_answer() frames from the inbound channel should be sent to all outbound channels instead of only if there is just one outbound channel. Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of the the outbound channels. This can happen if a blond transfer is done by a remote switch on the inbound channel. JIRA AST-443 JIRA SWP-2730 ........ * /, main/cli.c: Merged revisions 313366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) | 2 lines Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output. ........ 2011-04-11 19:39 +0000 [r313280] Leif Madsen * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 313279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313279 | lmadsen | 2011-04-11 14:36:40 -0500 (Mon, 11 Apr 2011) | 21 lines Merged revisions 313278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500 (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093) Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir (license 46) ........ ................ ................ 2011-04-11 15:47 +0000 [r313191] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 313190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines Merged revisions 313189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines Stuck channel using FEATD_MF if caller hangs up at the right time. The cause was actually a caller hanging up just at the end of the Feature Group D DTMF tones that setup the call. The reason for this is a "guard timer" that's implemented using ast_safe_sleep(100). If the caller happens to hang up AFTER the final tone of the DTMF string but BEFORE the end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero. This causes the code to bounce to the end of ss_thread(), but it does NOT tear down the call properly. This should be a rare occurrence because the caller has to hang up at EXACTLY the right time. Nonetheless, it was happening quite regularly on the reporter's system. It's not easily reproducible, unless you purposely increase the guard-time to 2000 or more. Once you do that, you can reproduce it every time by watching the DTMF debug and hanging up just as it ends. Simply add an ast_hangup() before goto quit. (closes issue #15671) Reported by: jcromes Patches: issue15671.patch uploaded by pabelanger (license 224) Tested by: jcromes ........ ................ ................ 2011-04-09 21:00 +0000 [r313143] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 313142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr 2011) | 3 lines fix trivial bug in ooh323_indicate on AST_CONTROL_SRC... check p->rtp is not null ........ 2011-04-08 16:17 +0000 [r313100] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h, channels/sig_pri.c, channels/sig_ss7.c: Add private lock deadlock avoidance callback to PRI and SS7. Factor out the equivalent function for analog. 2011-04-07 13:42 +0000 [r313049] Jonathan Rose * /, main/features.c: Merged revisions 313048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313048 | jrose | 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) | 16 lines Merged revisions 313047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | 9 lines Makes parking lots clear and rebuild properly when features reload is invoked from CLI Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared. (closes issue #18801) Reported by: mickecarlsson Review: https://reviewboard.asterisk.org/r/1161/ ........ ................ 2011-04-07 10:30 +0000 [r313003-313005] Alec L Davis * /, channels/sig_pri.c: Merged revisions 313001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines Fix ISDN calling subaddr User Specified Odd/Even Flag Calculation of the Odd/Even flag was wrong. Implement correct algo, and set odd/even=0 if data would be truncated. Only allow automatic calculation of the O/E flag, don't let dialplan influence. (closes issue #19062) Reported by: festr Patches: bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: festr, alecdavis, rmudgett ........ * apps/app_voicemail.c: app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE 2011-04-05 18:47 +0000 [r312868-312950] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 312949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines Crash if ISDN span layer 1 is down on initial load. Regression from -r312575 B channel shifting during negotiation. * Also combine updating the alarm flag with clearing the resetting flag. ........ * /, channels/chan_sip.c: Merged revisions 312889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) | 5 lines Add 416 response to OPTIONS packet. RFC3261 Section 11.2 says the response code to an OPTIONS packet needs to be the same as if it were an INVITE. ........ * /, channels/chan_sip.c: Merged revisions 312866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) | 15 lines Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension. The get_destination() function was not using the "s" extension when the request URI did not specify an extension. This is a regression caused when the URI parsing code was extracted into parse_uri(). Made get_destination() substitute the "s" extension when the parsed URI results in an empty string. (closes issue #18348) Reported by: shmaize Patches: issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested by: shmaize ........ 2011-04-05 14:16 +0000 [r312767] Matthew Nicholson * main/manager.c, /, configs/manager.conf.sample: Merged revisions 312766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 | mnicholson | 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) | 22 lines Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) | 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ ................ ................ 2011-04-05 13:55 +0000 [r312756] Jonathan Rose * apps/app_meetme.c: Minor change to 'L' option for meetme to include some verb statements for the option. 2011-04-04 19:31 +0000 [r312716] Richard Mudgett * channels/sig_pri.c: Remove the channel parameter from sig_pri_handle_subcmds(). It was only used in a debug message and may not be correct anyway. 2011-04-04 17:37 +0000 [r312678-312680] Jonathan Rose * pbx/pbx_config.c: In handle_cli_dialplan_add_extension, const char pointer *into_context is used instead of a->argv[5] to improve readability. * CHANGES, pbx/pbx_config.c: Makes 'dialplan add extension' create the specified context if it does not already exist. If the user invokes 'dialplan add extension' into a non-existing context, the context will be created and a message informing the user of the context being created will be issued in cli. (closes issue #17431) Reported by: leearcher Patches: context_auto_create.diff uploaded by kobaz (license 834) Tested by: leearcher, kobaz, jrose 2011-04-04 16:17 +0000 [r312579] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 312575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ ................ ................ 2011-04-01 23:17 +0000 [r312462-312510] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 312509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines When a call going out an NT-PTMP port gets rejected, Asterisk crashes. If a call is sent to an ISDN phone that rejects the call with RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes. I could not get my setup to crash. However, I could see the possibility from a race condition between queuing an AST_CONTROL_BUSY to the core and then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is processed before the AST_CONTROL_HANGUP is queued, the ast_channel could be destroyed out from under chan_misdn. Avoid this particular crash scenario by not queueing the AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy JIRA SWP-2679 ........ * /, main/ccss.c: Merged revisions 312461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail. The CallCompletionRequest()/CallCompletionCancel() dialplan applications exit nonzero on normal failure conditions. The nonzero exit causes the dialplan to hangup immediately. The dialplan author has no opportunity to report success/failure to the user. * Made always return zero so the dialplan can continue. * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also documented the values set. * Reduced the warning about no core instance in CallCompletionCancel() to a debug message. It is a normal event and should not be output at the WARNING level. (closes issue #18763) Reported by: p_lindheimer Patches: ccss.patch uploaded by p lindheimer (license 558) Modified Tested by: p_lindheimer, rmudgett JIRA SWP-3042 ........ 2011-04-01 17:28 +0000 [r312384-312423] Jonathan Rose * channels/chan_dahdi.c: Fixing bad line break from 312384 * channels/chan_dahdi.c, include/asterisk/dsp.h, CHANGES, main/dsp.c: New Feature for chan_dahdi. 4 length pattern matching. In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett 2011-04-01 10:59 +0000 [r312289] Tilghman Lesher * include/asterisk/select.h, /, addons/cdr_mysql.c, main/asterisk.c: Merged revisions 312286,312288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines Reload must react correctly against a possibly changed table, so dropping the conditional reload flag. ................ r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines Merged revisions 312287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) ........ ................ ................ 2011-04-01 09:08 +0000 [r312118-312212] Alec L Davis * /, apps/app_voicemail.c: Merged revisions 312211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines Merged revisions 312210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines voicemail: get real last_message_index and count_messages, ODBC resequence change last_message_index to read the max msgnum stored in the database change count_messages to actually count the number of messages. last_message_index change: This fixed overwriting of the last message if msgnum=0 was missing. Previously every incoming message would overwrite msgnum=1. count_messages change: allows us to detect when requencing is required in opneA_mailbox. resequence enabled for ODBC storage: Assists with fixing up corrupt databases with gaps, but only when a user actively opens there mailboxes. (closes issue #18692,#18582,#19032) Reported by: elguero Patches: based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37) Tested by: elguero, nivek, alecdavis Review: https://reviewboard.asterisk.org/r/1153/ ........ ................ ................ * /, apps/app_voicemail.c: Merged revisions 312117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines Merged revisions 312103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines app_voicemail: close_mailbox needs to respect additional messages while mailbox is open. close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ ........ ................ ................ 2011-03-31 20:12 +0000 [r311981-312023] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 312022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines chan_misdn segfaults when DEBUG_THREADS is enabled. The segfault happens because jb->mutexjb is uninitialized from the ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero value meant mutex tracking initialization had already happened. Recent changes to mutex tracking code to reduce excessive memory consumption exposed this uninitialized value. Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc(). Also eliminated redundant zero initialization code in the routine. (closes issue #18975) Reported by: irroot ........ * include/asterisk/channel.h: Fix function reference in comment. 2011-03-31 06:44 +0000 [r311931] Tilghman Lesher * /, configs/cdr_mysql.conf.sample: Merged revisions 311930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) | 6 lines Incorrect default example; the field is actually internally named "clid", not "callerid". (closes issue #19040) Reported by: wcselby Tested by: tilghman ........ 2011-03-30 01:57 +0000 [r311875] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 311874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) | 1 line Update some setup_dahdi_int() comments. ........ 2011-03-29 08:33 +0000 [r311806] Tilghman Lesher * cel/cel_odbc.c, /: Merged revisions 311799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) | 7 lines Remove extraneous check from integer-type fields. (closes issue #19027) Reported by: mlehner Review: https://reviewboard.asterisk.org/r/1149/ ........ 2011-03-28 22:00 +0000 [r311752] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 311751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 Mar 2011) | 2 lines Cross-reference VoiceMail() and VoiceMailMain() in the xml docs. ........ 2011-03-27 21:49 +0000 [r311688] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 311687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar 2011) | 2 lines correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS ........ 2011-03-23 21:55 +0000 [r311613-311616] Brett Bryant * /, apps/app_meetme.c: Merged revisions 311615 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (closes issue #18070) Reported by: mav3rick Review: https://reviewboard.asterisk.org/r/1132/ ........ * /, channels/sip/reqresp_parser.c: Merged revisions 311612 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null value. (closes issue #18821) Reported by: cmaj Patches: patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx uploaded by cmaj (license 830) ........ 2011-03-23 02:51 +0000 [r311559] Terry Wilson * /, channels/sip/reqresp_parser.c: Merged revisions 311558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines Don't use static declared buf in parse_name_andor_addr This function isn't used anywhere yet, but we definitely don't want to keep the same value for buf between calls to the function. ........ 2011-03-22 15:26 +0000 [r311498] David Vossel * /, apps/app_meetme.c: Merged revisions 311497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action ........ ................ 2011-03-18 19:05 +0000 [r311427] Jonathan Rose * CHANGES, apps/app_followme.c: Adds an option to FollowMe that isn't useful for the bug it was made to solve. Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost. 2011-03-18 16:27 +0000 [r311385] David Vossel * codecs/codec_resample.c: Remove libresample dependency from codec_resample.c 2011-03-18 16:24 +0000 [r311373] Jonathan Rose * /, channels/chan_sip.c, res/res_fax.c, res/res_jabber.c: Merged revisions 311352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL. This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings. (closes issue #18759) Reported by: bklang Patches: null-strings.patch uploaded by bklang (license 919) ........ 2011-03-18 16:03 +0000 [r311343] Matthew Nicholson * /, res/res_fax.c: Merged revisions 311342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar 2011) | 2 lines Properly populate the LOCALSTATIONID channel variable. ........ 2011-03-18 03:00 +0000 [r311296-311298] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 311297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines Race condition when ISDN CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. ........ * apps/app_dial.c, /: Merged revisions 311295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines Merged revision 310986 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines Dial() o option broke when connected line feature added. The patch restores the o option behavior and adds the ability to specify the CallerID. The Dial o and f options are complementary to each other. The o option stores the CallerID on the outgoing channel as the channel's CallerID. The f option forces the CallerID sent by the outgoing channel. o(x) - The argument 'x' is optional. If not present, then specify that the CallerID that was present on the *calling* channel be stored as the CallerID on the *called* channel. This was the behavior of Asterisk 1.0 and earlier. If present, then specify the CallerID stored on the *called* channel. Note that o(${CALLERID(all)}) is similar to option o without parameters. f(x) - The argument 'x' is optional and its presence changes the behavior of this option. If not present, then force the outgoing CallerID on a call-forward or deflection to the dialplan extension for this Dial() using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If present, then force the outgoing CallerID to 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA SWP-3096 .......... ................ 2011-03-17 19:05 +0000 [r311198] Jonathan Rose * /, apps/app_chanspy.c: Merged revisions 311197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy. (closes issue #18742) Reported by: jkister Tested by: jkister, jcovert, jrose Review: http://reviewboard.digium.internal/r/106/ ........ 2011-03-17 15:02 +0000 [r311142] Matthew Nicholson * main/manager.c, /: Merged revisions 311141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311141 | mnicholson | 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) | 11 lines Merged revisions 311140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) | 4 lines Don't write items to the manager socket twice. AST-2011-003 (closes issue 0018987) Reported by: ks-steven ........ ................ 2011-03-17 10:51 +0000 [r311051] Alec L Davis * /, configs/indications.conf.sample: Merged revisions 311050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311050 | alecdavis | 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) | 24 lines Merged revisions 311049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG ........ ................ ................ 2011-03-16 19:51 +0000 [r310941-311001] Terry Wilson * main/tcptls.c, /: Merged revisions 310999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310999 | twilson | 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) | 18 lines Merged revisions 310998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) | 11 lines Fix crash on fdopen failure See security advisory AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830) patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830) Tested by: cmaj, twilson ........ ................ * main/manager.c, /: Merged revisions 310993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310993 | twilson | 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) | 11 lines Merged revisions 310992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines Don't keep trying to write to a closed connection See security advisory AST-2011-003. ........ ................ * /, main/features.c: Merged revisions 310902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ ................ ................ 2011-03-15 01:49 +0000 [r310835] Tilghman Lesher * addons/chan_ooh323.c, /: Merged revisions 310834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 Mar 2011) | 2 lines Fix branch compile. ........ 2011-03-15 01:36 +0000 [r310833] Alec L Davis * /, main/utils.c: Merged revisions 310781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) | 10 lines core show locks: display ThreadID in hexadecimal Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) ........ 2011-03-14 21:51 +0000 [r310735] Alexandr Anikin * addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c, /, addons/ooh323c/src/ooCalls.h: Merged revisions 310734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 (closes issue #18693) ........ r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines Introduce t.38 parameters control functionality not full but enough for Send/RcvFax support Introduce t.38 controls between asterisk core and channel/proto layers. Not all parameters are transferred from proto layers but *Fax apps tested and work ok. (issue #18693) Reported by: benngard2 Patches: issue-18693.patch uploaded by may213 (license 454) ........ 2011-03-14 16:55 +0000 [r310637] Richard Mudgett * /, main/callerid.c: Merged revisions 310636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines Merged revisions 310635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek ........ ................ ................ 2011-03-14 15:40 +0000 [r310547-310588] Jonathan Rose * /, funcs/func_volume.c: Merged revisions 310587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines Merged revisions 310585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines Adds 'p' as an option to func_volume. When it is on, the old behavior with DTMF controlling volume adjustment will be enforced. When it is off, DTMF will not be processed by the function. Programmed by Jonathan Rose Reviewed by David Vossel, Leif Madsen, and Russell Bryant http://reviewboard.digium.internal/r/93/ ........ ................ * main/audiohook.c: Fixes null reference bug introduced by audio hook changes that affects various OS distributions. Thanks David. 2011-03-12 20:42 +0000 [r310416-310500] Tilghman Lesher * /, pbx/pbx_ael.c: Merged revisions 310462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310462 | tilghman | 2011-03-12 14:27:54 -0600 (Sat, 12 Mar 2011) | 45 lines Merged revisions 310448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines Add AELSub, which provides a stable entry point into AEL subroutines. This commit needs some explanation, given that we're adding a new application into an existing release branch. This is generally a violation of our release policy, except in very limited circumstances, and I believe this is one of those circumstances. The problem that this solves is one of the sanity of using multiple dialplan languages to define a dialplan. In the case of the reporter, he or she is using AEL is define subroutines, while using Realtime extensions to invoke those subroutines. While you can do this, it's based upon the reality of AEL using actual dialplan extensions; however, there is no guarantee that the details of _how_ AEL is compiled into extensions will remain stable. In fact, at the time of this commit, it has already changed twice, once in a fundamental way. Now normally, a new application would only be added to trunk. However, this application is explicitly to create a stable user-level API between versions, and adding it to trunk only will not solve the user's problem of switching between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10. Therefore, it needs to go into existing release branches. For the sake of consistency, and also because one of the changes was between 1.4 and 1.6.x, I am also electing to commit this to 1.4. (closes issue #18910) Reported by: alexandrekeller Patches: 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14) 20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14) Tested by: alexandrekeller ........ ................ ................ * /, funcs/func_odbc.c: Merged revisions 310415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310415 | tilghman | 2011-03-12 14:05:46 -0600 (Sat, 12 Mar 2011) | 14 lines Merged revisions 310414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) | 7 lines Transactional handles should be used for the insertbuf, if available. Also, fix a possible resource leak. (closes issue #18943) Reported by: irroot ........ ................ 2011-03-11 18:54 +0000 [r310373] Jonathan Rose * include/asterisk/audiohook.h, main/audiohook.c, CHANGES, apps/app_mixmonitor.c: Mix Monitor: Now with r and t options. 2011-03-11 15:09 +0000 [r310332] Kevin P. Fleming * Makefile, configure, codecs/gsm/Makefile, configure.ac, makeopts.in, codecs/lpc10/Makefile: Use "-march=native" when possible. Recent versions of GCC have a tuning option value of 'native', which causes the compiler to optimize the build for the CPU the compile is performed on. Since most people are building Asterisk on the machine they plan to run it on, the configure script and build system will now use this value unless a different value is specified by the user in CFLAGS when the configure script is executed. In addition, this value will be used for building the GSM and LPC10 codecs as well, in preference to the logic that has been in their Makefiles forever to optimize for certain types of CPUs. 2011-03-11 06:56 +0000 [r310288] Alec L Davis * main/rtp_engine.c, /: Merged revisions 310287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call If the channel condition is one of the following after breaking out of the loop, don't try to update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 Review: https://reviewboard.asterisk.org/r/1128/ ........ 2011-03-10 16:09 +0000 [r310241] Terry Wilson * main/manager.c, /, res/res_phoneprov.c: Merged revisions 310240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines Add \r\n to remaining http headers passed to ast_http_send r309204 changed the behavior of ast_http_send. It now requires headers to be passed with trailing \r\n. This change updates the remaining instances in the code that did not pass the \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama (license 1028) manager.diff.txt uploaded by twilson (license 396) Tested by: lathama ........ 2011-03-10 15:28 +0000 [r310238] Mark Michelson * /, channels/chan_sip.c: Merged revisions 310231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines Be more tolerant of what URI we accept for call completion PUBLISH requests. (closes issue #18946) Reported by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ 2011-03-10 05:54 +0000 [r310143] Tilghman Lesher * res/res_config_odbc.c, /, funcs/func_odbc.c, apps/app_voicemail.c: Merged revisions 310142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines Merged revisions 310141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ ................ 2011-03-08 20:34 +0000 [r310089] Jonathan Rose * /, channels/sip/dialplan_functions.c: Merged revisions 310088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines Returns with an error notice if CHANNEL function of SIP channel is read without arguments. (Closes issue #18653) Reported by: wuwu Patches: diff.patch uploaded by jrose (license 1225) Tested by: jrose ........ 2011-03-08 18:19 +0000 [r310045] Terry Wilson * /, res/res_calendar.c: Merged revisions 310039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) | 11 lines Spelling fix in "calendar show calendar" s/Cartegories/Catagories/ (closes issue #18931) Reported by: pdugas Patches: res_calendar.c.patch uploaded by pdugas (license 1222) Review: [full review board URL with trailing slash] ........ 2011-03-08 16:46 +0000 [r309996] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 309994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line Make pri parameter description consistent. ........ 2011-03-07 22:16 +0000 [r309859] Jonathan Rose * /, apps/app_mixmonitor.c: Merged revisions 309858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines Merged revisions 309857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: jrose ........ ................ ................ 2011-03-07 01:01 +0000 [r309809] Tilghman Lesher * channels/chan_dahdi.c, /, configure, include/asterisk/autoconfig.h.in, main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged revisions 309808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ 2011-03-07 00:14 +0000 [r309766] Mark Michelson * /, configs/sip.conf.sample: Merged revisions 309765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent. ........ 2011-03-05 17:53 +0000 [r309721] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 309720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines Fix caller id passed to openr2_chan_make_call (closes issue #18894) Reported by: malufrj Tested by: moy ........ 2011-03-05 10:30 +0000 [r309679] Tilghman Lesher * /, main/asterisk.c: Merged revisions 309678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines Merged revisions 309677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines Missed part of the conversion when we started passing ppid to astcanary. (closes issue #18850) Reported by: viraptor Patches: canary_ppid.patch uploaded by viraptor (license 543) ........ ................ 2011-03-04 23:22 +0000 [r309640] Terry Wilson * configs/calendar.conf.sample, include/asterisk/calendar.h, CHANGES, res/res_calendar.c: Add setvar option to calendaring Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ 2011-03-04 19:38 +0000 [r309493-309587] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 309585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309585 | mnicholson | 2011-03-04 13:38:25 -0600 (Fri, 04 Mar 2011) | 9 lines Merged revisions 309584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved. ........ ................ * /, pbx/pbx_lua.c: Merged revisions 309542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines Merged revisions 309541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file. Also, prepend a newline to traceback output so that the main error message is on it's own line. ........ ................ * /, pbx/pbx_lua.c: Merged revisions 309495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309495 | mnicholson | 2011-03-04 12:10:23 -0600 (Fri, 04 Mar 2011) | 9 lines Merged revisions 309494 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is never used ........ ................ * pbx/pbx_lua.c, configs/extensions.lua.sample: Add support for defining hints from pbx_lua (closes issue #16024) Reported by: mnicholson 2011-03-04 17:40 +0000 [r309491] Russell Bryant * channels/chan_nbs.c: Fix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk). In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants. 2011-03-04 16:00 +0000 [r309449] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 309448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar 2011) | 8 lines Export global symbols from pbx_lua to allow modules to be loaded. Fixes a regression introduced in r278132. (closes issue #18671) Reported by: Igels Patches: pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96) Tested by: Igels ........ 2011-03-04 15:28 +0000 [r309446] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, funcs/func_channel.c, channels/sig_pri.c, UPGRADE-1.8.txt: Merged revisions 309445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i/[:]- There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ 2011-03-04 01:52 +0000 [r309404] David Ruggles * /, apps/app_externalivr.c: Merged revisions 309403 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ ................ ................ 2011-03-02 21:08 +0000 [r309209-309300] Jason Parker * main/channel.c: Add HangupRequest manager event, to specify when/where a channel gets hung up. (closes issue #18226) Reported by: clegall_proformatique Patches: asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139) * /, channels/chan_sip.c: Merged revisions 309256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) ........ ................ * /, main/http.c: Merged revisions 309204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines Fix consistency of CRLFs on HTTP headers that get sent out. (closes issue #18186) Reported by: nivaldomjunior Patches: 18186-httpheadernewline.diff uploaded by qwell (license 4) ........ 2011-03-01 21:57 +0000 [r309127-309171] Richard Mudgett * /, funcs/func_channel.c: Merged revisions 309170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines Document CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Added XML documentation for CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML documentation for CHANNEL(reversecharge). ........ * channels/sig_analog.c, /: Merged revisions 309126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal. Looks like an unintended change when sig_analog.c was extracted from chan_dahdi.c. Removed useless conditional around needed code and fixed resulting compiler warning. (closes issue #18667) Reported by: enegaard Patches: issue18667.patch uploaded by enegaard (license 1197) Tested by: enegaard JIRA SWP-2965 ........ 2011-03-01 16:22 +0000 [r309090] David Vossel * /, channels/chan_sip.c: Merged revisions 309084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet ........ ................ 2011-02-28 11:16 +0000 [r308992-309036] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged revisions 309035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. ........ r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify meaning, removing double negative (stupid!) ........ ................ * /, funcs/func_odbc.c: Merged revisions 308991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines Merged revisions 308990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled. (closes issue #18815) Reported by: irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot (license 52) ........ ................ 2011-02-25 18:58 +0000 [r308946] Alec L Davis * /, channels/chan_sip.c: Merged revisions 308945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines Fix Deadlock with attended transfer of SIP call Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ 2011-02-24 21:43 +0000 [r308904] Richard Mudgett * main/channel.c, /: Merged revisions 308903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines Invalid read in ast_channel_set_caller_event(). Valgrind reported that ast_channel_set_caller_event() was reading data from a freed buffer when using the pre_set structure. Rearange things to pre-calculate the name and number pointer before updating the caller party structure to see if the name or number was changed. ........ 2011-02-24 17:59 +0000 [r308816] Terry Wilson * main/manager.c, /: Merged revisions 308815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines Merged revisions 308814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson ........ ................ ................ 2011-02-24 15:10 +0000 [r308724] Matthew Nicholson * main/udptl.c, /: Merged revisions 308723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines Merged revisions 308722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........ ................ ................ 2011-02-24 03:49 +0000 [r308680] Terry Wilson * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 308679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ 2011-02-23 23:55 +0000 [r308623-308624] Richard Mudgett * main/translate.c: Fix compiler warning. * /, channels/sig_pri.c: Merged revisions 308622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 ........ 2011-02-22 23:04 +0000 [r308582] David Vossel * main/format.c, funcs/func_speex.c, main/frame.c, main/rtp_engine.c, include/asterisk/silk.h (added), codecs/speex/fixed_generic.h (added), bridges/bridge_softmix.c, channels/chan_gtalk.c, bridges/bridge_multiplexed.c, channels/chan_iax2.c, main/format_pref.c, codecs/speex/resample.c (added), main/channel.c, funcs/func_pitchshift.c, include/asterisk/audiohook.h, channels/chan_skinny.c, main/format_cap.c, funcs/func_volume.c, codecs/speex (added), codecs/codec_resample.c, include/asterisk/format.h, codecs/speex/arch.h (added), include/asterisk/frame.h, include/asterisk/rtp_engine.h, codecs/speex/stack_alloc.h (added), main/bridging.c, apps/app_jack.c, configs/codecs.conf.sample, res/res_rtp_asterisk.c, formats/format_attr_silk.c (added), channels/chan_sip.c, main/translate.c, main/slinfactory.c, codecs/codec_speex.c, include/asterisk/_private.h, CHANGES, codecs/speex/speex_resampler.h (added), res/res_mutestream.c, include/asterisk/format_cap.h, codecs/Makefile, channels/chan_jingle.c, main/data.c, channels/iax2.h, main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, main/asterisk.c, include/asterisk/slinfactory.h, include/asterisk/translate.h, codecs/speex/resample_sse.h (added), include/asterisk/time.h: Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ 2011-02-22 15:33 +0000 [r308527] Andrew Latham * main/http.c: Use ast_debug for console logging Guessed the log levels based on info that level 3 is the soft roof. Can we create a page / document to define the levels? 2011-02-21 15:04 +0000 [r308417] Matthew Nicholson * main/udptl.c, /: Merged revisions 308416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines Merged revisions 308414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 ........ ................ ................ 2011-02-21 14:14 +0000 [r308372] Andrew Latham * main/http.c: Add HTTP URI Debug logging and update notice enable reporting of the request URI / URL in debugging change funny debug note to a serious note. 2011-02-21 13:58 +0000 [r308371] Tzafrir Cohen * main/pbx.c: fix a memory leak in device state The callback handle_statechange (pbx.c) fails to release its data pointer, leaking memory in the process. Reported by: tzafrir Patches: 18735_pbx_free_callback.diff uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/1110/ 2011-02-19 14:07 +0000 [r308331] Andrew Latham * main/http.c: Add CSS MIME Type Modern browsers are checking for the MIME Type of pages and in some cases will not load a file if the type is wrong. 2011-02-19 11:03 +0000 [r308289] Tilghman Lesher * utils, /: Merged revisions 308288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) | 2 lines A few more (copies of) files to ignore in this directory. ........ 2011-02-18 00:11 +0000 [r308243] Alexandr Anikin * addons/chan_ooh323.c, /, addons/ooh323cDriver.c, addons/ooh323cDriver.h: Merged revisions 308242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines added g729onlyA option for announce only AnnexA g.729 codec in h.323 capabilities. Option can be global or per user/peer. ........ 2011-02-17 20:21 +0000 [r308205] Richard Mudgett * channels/chan_dahdi.c: Add more verbage to CLI command 'pri show channels' usage. 2011-02-16 22:02 +0000 [r308157] Paul Belanger * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines Fix FreeBSD builds. ........ 2011-02-16 08:06 +0000 [r308099] Alexandr Anikin * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines ifdef __linux__ keepalive variables also ........ 2011-02-15 23:34 +0000 [r308013] Jason Parker * /, apps/app_queue.c: Merged revisions 308010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ ................ 2011-02-15 23:07 +0000 [r307969] Alexandr Anikin * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls only on linux, freebsd and others don't have these options on sockets. 2011-02-15 21:42 +0000 [r307963-307964] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: Add CLI "pri show channels" command. List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 * apps/app_dial.c, /: Merged revisions 307962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line Don't crash when forcing caller id. ........ 2011-02-15 18:09 +0000 [r307927] David Vossel * channels/chan_phone.c: Fixes compile error in chan_phone for big endian 2011-02-15 16:18 +0000 [r307883] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/chan_sip.c, main/ccss.c, channels/sig_pri.c, include/asterisk/ccss.h: Merged revisions 307879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ 2011-02-15 07:03 +0000 [r307751-307838] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 307837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307837 | tilghman | 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) | 15 lines Merged revisions 307836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines Need to retrieve the rows affected before using the associated variable. (closes issue #18795) Reported by: irroot Patches: 20110211__issue18795.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ * /, res/res_odbc.c: Merged revisions 307793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307793 | tilghman | 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) | 15 lines Merged revisions 307792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) | 8 lines Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once. (issue #18156) Reported by: asgaroth Patches: 20110214__issue18156.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ * addons/chan_ooh323.c, addons/ooh323c/src/ooCmdChannel.c: Making trunk compile again. * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 307750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm ........ 2011-02-13 10:50 +0000 [r307677-307713] Alexandr Anikin * addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooCmdChannel.c: lc not found - it's warning, not error, change malloc to ast_calloc again * addons/chan_ooh323.c, addons/ooh323cDriver.c: change malloc to ast_calloc calls to prevent crash of asterisk 2011-02-10 22:43 +0000 [r307537] Jason Parker * contrib/init.d/rc.debian.asterisk, /, main/asterisk.c: Merged revisions 307536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines Merged revisions 307535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines Remove color when executing commands via a remote console. Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'm reverting here. (issue #18776) Reported by: alecdavis ........ ................ ................ 2011-02-10 17:45 +0000 [r307468] Mark Michelson * /, configs/ccss.conf.sample: Merged revisions 307467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines Fix a gaffe in the CCSS sample configuration. Discovered by Philippe Lindheimer and pointed out on #asterisk-dev ........ 2011-02-10 17:12 +0000 [r307433] David Vossel * channels/chan_sip.c, main/format_cap.c, include/asterisk/format_cap.h: Fixes bug in chan_sip where nativeformats are not set correctly. The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. 2011-02-10 13:29 +0000 [r307396] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c: Corrections for properly work with H.323v2 (older) endpoints and other small fixes. Interpret remote side H.225 version. Corrections for H.323v2 endpoints: don't start TCS and MSD before connect, don't start TCS and MSD by accepting H.245 connection, start TCS and MSD by StartH245 facility message. Other fixes: fix non zeroended remoteDisplayName issue, small fixes in call clearing by closing H.245 connection, tcp keepalive introduced on TCP connections (now is hardcoded, will be configurable in the future), don't force H.245tunneling if FastStart is active, don't send Alerting singal more than once per call. (closes issue #18542) Reported by: vmikhelson Patches: issue18542-final-3.patch uploaded by may213 (license 454) Tested by: vmikhelson 2011-02-09 22:48 +0000 [r307359] Jeff Peeler * apps/app_meetme.c, CHANGES: Add new manager action MeetmeListRooms. From the submitter: I've added a new manager action to list only the active conferences on an Asterisk system. It shows the same data displayed when you run a 'meetme list' on the Asterisk CLI. (closes issue #17905) Reported by: rcasas Patches: app_meetme.c.patch uploaded by rcasas (license 641) Review: https://reviewboard.asterisk.org/r/874/ 2011-02-09 21:46 +0000 [r307315] Andrew Latham * contrib/init.d/rc.debian.asterisk: Disable color during running test (closes issue #18776) Reported by: alecdavis Patches: ast_deb_init.diff uploaded by lathama (license 1028) Tested by: andrel, lathama 2011-02-09 21:08 +0000 [r307229-307274] Jeff Peeler * /, main/astobj2.c: Merged revisions 307273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) | 8 lines Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback. (closes issue #18758) Reported by: rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) ........ * main/features.c, CHANGES: Allow parkedmusicclass to be settable for non-default parking lots. (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff * /, main/features.c: Merged revisions 307228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines Make sure to set parking dial context for non-default parking lots. Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me ........ ................ 2011-02-09 19:17 +0000 [r307192] Tzafrir Cohen * main/loader.c: clarify warning when no loadable module support Clarify warning message when LOADABLE_MODULES is disabled but we still try to load a module. 2011-02-09 05:53 +0000 [r307143] Tilghman Lesher * main/lock.c, /: Merged revisions 307142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) | 3 lines Initialize tracking variable in structure properly. Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by me.) ........ 2011-02-08 21:24 +0000 [r307097] Jason Parker * /, main/logger.c: Merged revisions 307092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines Fix issue with verbose messages not showing on remote console. This code was reworked recently, and since the logchannel list hadn't been created yet at this point, and it was a verbose message, it was being dropped on the floor. Now it'll continue on to where it should be handled. (closes issue #18580) Reported by: pabelanger ........ 2011-02-08 21:18 +0000 [r307071] Mark Michelson * /, main/ccss.c: Merged revisions 307065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines Add a couple of useful channel variables for the CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine the channel and context that will be called when the recall occurs. ........ 2011-02-08 20:42 +0000 [r307061] Terry Wilson * /, channels/chan_sip.c: Merged revisions 306979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306979 | twilson | 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ ................ ................ 2011-02-08 20:31 +0000 [r307041] Andrew Latham * /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample, doc/asterisk.sgml: Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir 2011-02-08 19:42 +0000 [r306867-306968] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 306967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306967 | jpeeler | 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line fix this line again ........ ................ ................ * /, apps/app_voicemail.c: Merged revisions 306962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines Merged revisions 306961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil ........ ................ ................ * /, apps/app_voicemail.c: Merged revisions 306866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306866 | jpeeler | 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line make this safer and fully correct, pointed out by Steve Davis ........ ................ ................ 2011-02-08 02:05 +0000 [r306827] Andrew Latham * doc/asterisk.sgml: Documentation Updates. Start updates to the man pages. (issue #16505) Reported by: tzafrir Tested by: lathama 2011-02-08 00:43 +0000 [r306755-306793] Richard Mudgett * configs/chan_dahdi.conf.sample: Define the MCID acronym in chan_dahdi.conf.sample. * channels/sig_pri.h: Use correct conditional for MCID send. * channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, funcs/func_frame_trace.c, main/features.c, CHANGES, channels/sig_pri.c, include/asterisk/frame.h: Pass a MCID request to the bridged channel. Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 2011-02-07 22:46 +0000 [r306670-306675] Terry Wilson * /, main/features.c: Merged revisions 306674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines Merged revisions 306673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 306619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines Merged revisions 306618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ ................ ................ 2011-02-07 17:55 +0000 [r306576] Mark Michelson * /, main/ccss.c: Merged revisions 306575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines Rearrange a bit of code in the generic CC recall operation. By waiting to call the callback macro after the CC_INTERFACES, extension, priority, and context have been set, this information can be accessed more easily within the callback macro. Reported by Philippe Lindheimer. ........ 2011-02-07 16:33 +0000 [r306541] David Vossel * channels/chan_sip.c: Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary. 2011-02-05 22:16 +0000 [r306499] Alexandr Anikin * addons/chan_ooh323.c: fix trivial issue after dvossel patch, initial zero fill user and peer structure before cap structure allocated. 2011-02-05 02:55 +0000 [r306464] Richard Mudgett * channels/chan_dahdi.c: Ignore voice frames in chan_dahdi native bridging. Hardware is handling them. 2011-02-04 22:37 +0000 [r306432] Jeff Peeler * main/manager.c: Send manager event for blackfilter only if it DOES NOT match. The logic got reversed, oops. Works properly now when multiple blackfilters are present. (closes issue #18283) Reported by: telecos82 Patches: ast_managereventfilter.patch uploaded by telecos82 (license 687) 2011-02-04 20:30 +0000 [r306396] Richard Mudgett * apps/app_dial.c, channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, CHANGES, channels/sig_pri.c: Add ISDN display ie text handling options to chan_dahdi.conf. The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 2011-02-04 19:24 +0000 [r306359] Jason Parker * /, apps/app_queue.c: Merged revisions 306356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) ........ ................ 2011-02-04 19:09 +0000 [r306325-306326] Richard Mudgett * tests/test_format_api.c: Fix compiler warning. * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 306324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines Don't send redirecting updates to the caller if the dialplan forked the call. Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. ........ 2011-02-04 18:16 +0000 [r306258-306292] Paul Belanger * utils/extconf.c: Revert changes to extconf.c It seems extconf.c already defines some local ast_debug() functions. Theses should be removed and replaced with logger.h. A patch will be added to reviewboard shortly. * cel/cel_radius.c, addons/chan_ooh323.c, apps/app_meetme.c, main/say.c, channels/chan_gtalk.c, main/taskprocessor.c, res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c, pbx/pbx_loopback.c, main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c, res/res_smdi.c, channels/chan_skinny.c, main/features.c, res/res_agi.c, main/http.c, main/logger.c, res/ais/evt.c, main/app.c, res/res_config_ldap.c, apps/app_rpt.c, res/res_rtp_asterisk.c, main/pbx.c, channels/chan_sip.c, apps/app_fax.c, include/asterisk/channel.h, channels/sig_pri.c, channels/chan_misdn.c, include/asterisk/sched.h, utils/extconf.c, codecs/codec_ilbc.c, main/audiohook.c, res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c: Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ 2011-02-04 16:42 +0000 [r306257] David Vossel * codecs/codec_ilbc.c, codecs/ex_ilbc.h: Fix compile error in codec ilbc translator. 2011-02-03 23:50 +0000 [r306216] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 306215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines Fix SIP deadlock involving state changes. Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) has caused locking problems. Both of these functions lock the channel when the channel argument is passed in! In this case, the suspected problem (the backtrace makes it impossible to tell) was the private being locked in sip_set_rtp_peer and then: transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to verify that the fix was only required in 1.8 and later.) (closes issue #18491) Reported by: cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830) Tested by: cmaj ........ 2011-02-03 21:13 +0000 [r306128] Terry Wilson * channels/chan_local.c, /: Merged revisions 306127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ ................ ................ 2011-02-03 20:51 +0000 [r306125] Jeff Peeler * /, main/features.c: Merged revisions 306124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306124 | jpeeler | 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) | 17 lines Merged revisions 306123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines Set exception on channel in parking thread when POLLPRI event detected. This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) ........ ................ 2011-02-03 18:37 +0000 [r306086] Jason Parker * main/frame.c: Modify alignment of 'core show codecs', since the ID is no longer a huge int. 2011-02-03 18:12 +0000 [r306010-306053] David Vossel * main/frame.c: Fixes output of "core show codecs" to display image types correctly. * apps/app_dahdibarge.c, channels/chan_local.c, main/frame.c, apps/app_record.c, apps/app_alarmreceiver.c, bridges/bridge_softmix.c, formats/format_sln16.c, apps/app_ices.c, bridges/bridge_multiplexed.c, channels/chan_iax2.c, main/astobj2.c, res/res_rtp_multicast.c, channels/chan_dahdi.c, include/asterisk/bridging_technology.h, funcs/func_pitchshift.c, pbx/pbx_spool.c, include/asterisk/audiohook.h, channels/chan_skinny.c, channels/sip/include/globals.h, apps/app_dumpchan.c, formats/format_pcm.c, formats/format_h263.c, main/bridging.c, codecs/ex_ulaw.h, channels/sip/include/sip.h, main/pbx.c, codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c, formats/format_ogg_vorbis.c, bridges/bridge_simple.c, include/asterisk/channel.h, apps/app_talkdetect.c, channels/iax2-parser.c, include/asterisk/format_cap.h (added), apps/app_speech_utils.c, channels/iax2-parser.h, main/data.c, funcs/func_channel.c, main/audiohook.c, codecs/codec_dahdi.c, include/asterisk/frame_defs.h, formats/format_g726.c, apps/app_mixmonitor.c, main/asterisk.c, res/res_calendar.c, apps/app_voicemail.c, channels/chan_vpb.cc, addons/format_mp3.c, formats/format_sln.c, apps/app_dictate.c, codecs/ex_g722.h, codecs/codec_gsm.c, codecs/ex_g726.h, channels/chan_gtalk.c, include/asterisk/abstract_jb.h, main/channel.c, apps/app_mp3.c, codecs/codec_resample.c, formats/format_h264.c, formats/format_siren14.c, apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, channels/chan_sip.c, codecs/ex_lpc10.h, include/asterisk/format_pref.h (added), codecs/codec_alaw.c, res/res_adsi.c, tests/test_format_api.c (added), apps/app_originate.c, channels/chan_jingle.c, formats/format_vox.c, main/abstract_jb.c, include/asterisk/bridging.h, main/callerid.c, main/file.c, apps/app_sms.c, formats/format_g723.c, main/dsp.c, main/format.c (added), main/udptl.c, main/rtp_engine.c, addons/chan_ooh323.c, codecs/codec_adpcm.c, apps/app_test.c, addons/chan_ooh323.h, include/asterisk/speech.h, codecs/ex_adpcm.h, codecs/ex_alaw.h, formats/format_wav_gsm.c, include/asterisk/data.h, codecs/ex_gsm.h, main/indications.c, main/format_pref.c (added), main/cli.c, main/features.c, include/asterisk/mod_format.h, apps/app_amd.c, addons/ooh323cDriver.c, channels/chan_alsa.c, formats/format_jpeg.c, addons/ooh323cDriver.h, formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, formats/format_g719.c, channels/h323/ast_h323.cxx, channels/chan_bridge.c, apps/app_echo.c, apps/app_fax.c, codecs/codec_speex.c, include/asterisk/slin.h, channels/chan_agent.c, channels/iax2-provision.c, codecs/ex_speex.h, channels/chan_misdn.c, include/asterisk/image.h, channels/iax2.h, codecs/codec_ilbc.c, apps/app_chanspy.c, res/res_fax_spandsp.c, include/asterisk/slinfactory.h, include/asterisk/translate.h, channels/chan_unistim.c, channels/chan_multicast_rtp.c, main/ccss.c, apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c, formats/format_siren7.c, formats/format_ilbc.c, include/asterisk/file.h, include/asterisk/callerid.h, channels/chan_phone.c, main/dial.c, main/manager.c, main/format_cap.c (added), funcs/func_frame_trace.c, res/res_agi.c, main/app.c, apps/app_confbridge.c, include/asterisk/format.h (added), main/image.c, include/asterisk/rtp_engine.h, include/asterisk/frame.h, addons/chan_mobile.c, apps/app_parkandannounce.c, apps/app_jack.c, res/res_clioriginate.c, res/res_rtp_asterisk.c, apps/app_nbscat.c, codecs/codec_a_mu.c, res/res_fax.c, apps/app_festival.c, apps/app_waitforsilence.c, include/asterisk/astobj2.h, main/slinfactory.c, main/translate.c, channels/chan_console.c, channels/h323/chan_h323.h, channels/chan_oss.c, channels/chan_usbradio.c, channels/chan_h323.c, codecs/codec_ulaw.c, include/asterisk/pbx.h, channels/chan_nbs.c, formats/format_g729.c: Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ 2011-02-03 16:13 +0000 [r305988] Andrew Latham * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample: res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support (issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama 2011-02-03 00:29 +0000 [r305939] Richard Mudgett * main/channel.c, main/manager.c, /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 305923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ 2011-02-02 20:06 +0000 [r305845] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 305844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) | 5 lines Eliminate a file descriptor leak when using the FILE() dialplan function. (closes issue #18731) Reported by: marioabajo ........ 2011-02-02 19:30 +0000 [r305759-305843] Andrew Latham * configs/iax.conf.sample, funcs/func_enum.c, configs/dundi.conf.sample, funcs/func_callcompletion.c, /, configs/mgcp.conf.sample, configs/iaxprov.conf.sample, configs/unistim.conf.sample, apps/app_externalivr.c, configs/sip.conf.sample, configs/skinny.conf.sample, configs/h323.conf.sample, configs/sla.conf.sample, apps/app_voicemail.c: Replacing doc/* and asterisk.pdf with wiki links Adding links to http(s)://wiki.asterisk.org * configs/chan_dahdi.conf.sample, /, configs/extconfig.conf.sample, configs/res_snmp.conf.sample, main/ast_expr2f.c, res/res_timing_dahdi.c, configs/ccss.conf.sample, configs/sip.conf.sample, configs/skinny.conf.sample, main/config.c, configs/h323.conf.sample, configs/sla.conf.sample, main/ast_expr2.fl, res/res_srtp.c: Replacing doc/* with wiki links Adding links to http(s)://wiki.asterisk.org * /, channels/chan_sip.c: Replace link to old doc with new wiki page. Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions 2011-02-01 22:48 +0000 [r305693] Jason Parker * /, channels/chan_iax2.c: Merged revisions 305692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) | 7 lines Reverse sense of an error test when reading from astdb. (closes issue #18545) Reported by: jcovert Patches: chan_iax2.c.patch uploaded by jcovert (license 551) ........ 2011-02-01 21:16 +0000 [r305650] Andrew Latham * configs/sip.conf.sample: SIP Configuration Documentation sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. 2011-02-01 19:27 +0000 [r305604] Brett Bryant * cel/cel_pgsql.c, /: Merged revisions 305603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305603 | bbryant | 2011-02-01 14:23:20 -0500 (Tue, 01 Feb 2011) | 4 lines Add a possible solution to a customer problem with reloading cel_pgsql.so quickly. ........ 2011-02-01 18:03 +0000 [r305561] Andrew Latham * /: doc/tex dir removed, but corresponding entries still exists Update README, CHANGES, and Makefile. Direct users to http://wiki.asterisk.org for documentation or to the AST.txt and AST.pdf included in the tarball. (closes issue #18443) Reported by: bas Patches: changes.diff uploaded by lathama (license 1028) readme.diff uploaded by lathama (license 1028) Tested by: lathama bas 2011-02-01 17:05 +0000 [r305474] Jason Parker * /, res/res_musiconhold.c: Merged revisions 305473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305473 | qwell | 2011-02-01 11:04:23 -0600 (Tue, 01 Feb 2011) | 23 lines Merged revisions 305472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305472 | qwell | 2011-02-01 11:02:09 -0600 (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) | 9 lines Close file descriptor for timing source when a MOH class gets destroyed. (closes issue #18457) Reported by: mcallist Patches: 18457-closetimer.diff uploaded by qwell (license 4) 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested by: qwell, loloski ........ ................ ................ 2011-02-01 16:05 +0000 [r305433] Brett Bryant * apps/app_confbridge.c: Add's two features to confbridge: confbridge kick, and confbridge list. (closes issue #14389) (closes issue #18007) Reported by: jcollie Patches: 0001-Fix-up-bridging-module-so-that-menuselect-works.patch uploaded by jcollie (license 412) 0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch uploaded by jcollie (license 412) Tested by: file Review: https://reviewboard.asterisk.org/r/1084/ 2011-02-01 00:07 +0000 [r305344] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 305343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ ................ ................ 2011-01-31 23:08 +0000 [r305132-305255] Jason Parker * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ ................ ................ * main/tcptls.c, /, configs/sip.conf.sample: Merged revisions 305247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines Add alternative name for config option. The SIP sample configuration had "tlscadir" as the option name, but chan_sip used the more correct "tlscapath". Now both are accepted. Discovered (sort of) by a user on IRC in #asterisk ........ * /, res/res_musiconhold.c: Merged revisions 305198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305198 | qwell | 2011-01-31 15:30:44 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compile error. pseudofd no longer exists. ........ * /, res/res_musiconhold.c: Merged revisions 305131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305131 | qwell | 2011-01-31 15:00:25 -0600 (Mon, 31 Jan 2011) | 16 lines Merged revisions 305130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305130 | qwell | 2011-01-31 14:59:37 -0600 (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan 2011) | 2 lines Set file descriptors to -1 on creation, so that we don't see weirdness later. ........ ................ ................ 2011-01-31 13:57 +0000 [r305084] Andrew Latham * main/http.c: Asterisk HTTP response Content-type Address content type for BSD and other platforms (closes issue #18456) Reported by: alexo Patches: asterisk18_http.patch uploaded by alexo (license 1175) Tested by: alexo 2011-01-31 07:52 +0000 [r304951-305041] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 305040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305040 | tilghman | 2011-01-31 01:51:40 -0600 (Mon, 31 Jan 2011) | 2 lines Use the non-specific API aliases, to avoid a problem with building the utils directory. ........ * /, apps/app_voicemail.c: Merged revisions 304985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304985 | tilghman | 2011-01-31 01:27:13 -0600 (Mon, 31 Jan 2011) | 16 lines Merged revisions 304978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600 (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. ........ ................ ................ * main/lock.c, /, main/heap.c, main/utils.c, include/asterisk/lock.h, .cleancount: Merged revisions 304950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) | 18 lines Change mutex tracking so that it only consumes memory in the core mutex object when it's actually being used. This reduces the overall size of a mutex which was 3016 bytes before this back down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex). The exactness of the numbers here may vary slightly based upon how mutexes are implemented on a platform, but the long and short of it is that prior to this commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more than a table of 32767 locks. After this commit, the same table occupies a mere 7MB of memory. (closes issue #18194) Reported by: job Patches: 20110124__issue18194.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/1066 ........ 2011-01-30 00:22 +0000 [r304913] Andrew Latham * funcs/func_callcompletion.c, /, apps/app_externalivr.c, apps/app_queue.c, apps/app_voicemail.c, funcs/func_realtime.c, res/res_calendar.c: Add Function and Application Relationships to documentation Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. 2011-01-29 23:10 +0000 [r304639-304867] Sean Bright * /, res/res_config_ldap.c: Merged revisions 304866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304866 | seanbright | 2011-01-29 18:07:18 -0500 (Sat, 29 Jan 2011) | 14 lines Merged revisions 304865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat, 29 Jan 2011) | 7 lines Plug some memory leaks in the LDAP realtime driver. (closes issue #18435) Reported by: zaltar Patches: res_config_ldap.patch uploaded by zaltar (license 1148) ........ ................ * /, apps/app_meetme.c: Merged revisions 304777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500 (Sat, 29 Jan 2011) | 22 lines Merged revisions 304776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) | 15 lines If we fail to allocate our announcement objects, make sure we don't leak objects. The majority of this patch was committed already in r304726 and r304729. (issue #18225) Reported by: kenji (issue #18444) Reported by: junky (closes issue #18343) Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz (license 834) ........ ................ * /, apps/app_meetme.c: Merged revisions 304774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines Merged revisions 304773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines When we pass the S() or L() options to MeetMe, make sure that we honor C as well. Without this patch, if the user was kicked from the conference via the S() or L() mechanism, we would just hang up on them even if we also passed C (continue in dialplan when kicked). With this patch we honor the C flag in those cases. (closes issue #17317) Reported by: var ........ ................ * /, apps/app_meetme.c: Merged revisions 304730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines Merged revisions 304729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines Make sure that we unref the correct object when ejecting the most recent caller. Currently, when we kick the last user to enter, we decrement our own reference count which results in a crash when we kick another user or when we exit the conference ourselves. This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in 1.6.2. (closes issue #18225) Reported by: kenji Patches: issue18225.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ * /, apps/app_meetme.c: Merged revisions 304727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines Merged revisions 304726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines Fix user reference leak in MeetMe. We were unlinking the user from the conferences user container, but not decrementing the reference count of the user as well, resulting in a leak. (closes issue #18444) Reported by: junky Tested by: seanbright ........ ................ * /, apps/app_meetme.c: Merged revisions 304683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines Merged revisions 304659,304682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines Don't leak references if we can't create a pseudo channel for mixing in MeetMe. If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. ........ r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous commit that snuck in. ........ ................ * /, main/acl.c: Merged revisions 304638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan 2011) | 11 lines Restore some conditionals that we lost in r277814. There are some cases where ast_append_ha() is called with a NULL instead of a valid int pointer. So if we get a NULL, don't try to dereference it. (closes issue #18162) Reported by: imcdona Patches: issue0018162.patch uploaded by pabelanger (license 224) Tested by: enegaard ........ 2011-01-27 20:09 +0000 [r304600] Brett Bryant * res/res_config_pgsql.c: Patch that fixes the "realtime show pgsql cache" command crash when giving a table name, because of the use of an uninitialized variable. Fixes an error introduced in r300882. (closes issue #18605) Reported by: romain_proformatique Patches: res_config_pgsql_fix.patch uploaded by romain proformatique (license 975) Tested by: romain_proformatique 2011-01-27 20:07 +0000 [r304599] Kevin P. Fleming * res/res_fax.c: Fix bug with 'F' option for ReceiveFAX and SendFAX. Skipping the call to set_t38_fax_caps() caused the FAX session details to not be marked as supporting audio FAX either... the function's name is a bit misleading. This patch restores the single bit of non-T.38 behavior from that function when audio mode is forced. 2011-01-27 19:12 +0000 [r304555] Richard Mudgett * /, main/ccss.c: Merged revisions 304554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011) | 4 lines Warning message if CALLCOMPLETION(cc_callback_macro or cc_agent_dialstring) are empty. Test if the value pointer is not NULL instead of not ast_strlen_zero(). ........ 2011-01-27 17:03 +0000 [r304463-304467] Jason Parker * /, configure, configure.ac: Merged revisions 304466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304466 | qwell | 2011-01-27 11:03:01 -0600 (Thu, 27 Jan 2011) | 23 lines Merged revisions 304465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304465 | qwell | 2011-01-27 11:01:24 -0600 (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | 9 lines Fix default prefix=/usr regression on non-Linux systems. This partially reverts a change made in branches/1.4/ r267759, which will cause issue #17013 to be reopened. This issue was pointed out by a user on #asterisk, who helpfully discovered that paths were being set incorrectly. To truly understand what was wrong, one should run: svn diff --force -c configure ........ ................ ................ * /, configure: Merged revisions 304462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304462 | qwell | 2011-01-27 10:48:44 -0600 (Thu, 27 Jan 2011) | 16 lines Merged revisions 304461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304461 | qwell | 2011-01-27 10:48:00 -0600 (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is more obvious what my next commit changes. ........ ................ ................ 2011-01-27 15:57 +0000 [r304422] Kevin P. Fleming * res/res_fax.c: Rename the SendFAX/ReceiveFAX 'force audio' option. The recently added option to disable usage of T.38 for a single session should have been named 'F' for 'force audio', since that is really what the user is asking to happen (and it's a positive option instead of a negative option that way). 2011-01-27 00:06 +0000 [r304385] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, channels/sig_pri.c: Merged from revision 304341 https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf pricpndialplan option. * Added from_channel value to prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... 2011-01-26 23:41 +0000 [r304384] Jeff Peeler * apps/app_followme.c: Add option to followme to delay answer until ready to bridge call. Followme answers an incoming call if it hasn't already been answered and starts MOH. Some poorly designed autodialers see the answer and start playing their message to the hold music. The 'N' option has been added to indicate ringing and not answer until the call is accepted. (closes issue #18479) Reported by: ianc Patches: trunk_followme.diff uploaded by ianc (license 998) 2011-01-26 22:39 +0000 [r304342] Kevin P. Fleming * res/res_fax.c: Add ability to disable T.38 usage for specific SendFAX/ReceiveFAX sessions. Sometimes during troubleshooting it can be useful to disable T.38 usage in order to narrow down a problem. This patch adds an 'n' option to SendFAX and ReceiveFAX so that can be done without having to disable T.38 usage entirely for the peer that Asterisk is communicating with. (inspired by trying to assist Bryant Zimmerman on asterisk-users) 2011-01-26 22:27 +0000 [r304340] Jeff Peeler * /, main/features.c: Merged revisions 304339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304339 | jpeeler | 2011-01-26 16:27:30 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011) | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. ........ ................ 2011-01-26 21:03 +0000 [r304252] Mark Michelson * main/udptl.c, /: Merged revisions 304250 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl ........ ................ 2011-01-26 20:44 +0000 [r304246] Matthew Nicholson * main/netsock2.c, /, channels/chan_sip.c, channels/sip/reqresp_parser.c, include/asterisk/netsock2.h, channels/sip/include/sip.h, channels/sip/include/reqresp_parser.h: Merged revisions 304245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ ................ 2011-01-26 20:25 +0000 [r304195] Sean Bright * /, configs/queues.conf.sample: Merged revisions 304186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304186 | seanbright | 2011-01-26 15:23:48 -0500 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304181 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan 2011) | 1 line Make sure the sample queues.conf is properly commented. ........ ................ ................ 2011-01-26 19:58 +0000 [r304152] Matthew Nicholson * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions 303907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan 2011) | 2 lines Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. ........ 2011-01-26 19:40 +0000 [r304151] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 304150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... ................ ................ 2011-01-26 01:27 +0000 [r304098] Sean Bright * /, main/file.c: Merged revisions 304097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304097 | seanbright | 2011-01-25 20:26:26 -0500 (Tue, 25 Jan 2011) | 19 lines Merged revisions 304096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines Per the man page, setvbuf() must be called before any other operation on an open file. We use setvbuf() to associate a buffer with a stream, but we have already written to the open file. This works (by chance) on Linux, but fails on other platforms, such as OpenSolaris. (closes issue #16610) Reported by: bklang Patches: setvbuf.patch uploaded by crjw (license 963) Tested by: bklang, asgaroth, efutch ........ ................ 2011-01-25 23:31 +0000 [r304008] Richard Mudgett * /, main/features.c: Merged revisions 304007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304007 | rmudgett | 2011-01-25 17:28:25 -0600 (Tue, 25 Jan 2011) | 22 lines Merged revisions 304006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines DTMF attended transfers sometimes fail for no apparent reason. The loop in feature_request_and_dial() can exit when Party C has answered without processing an AST_CONTROL_ANSWER. Also sometimes an AST_CONTROL_ANSWER never happens even though Party C has answered. Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER. ........ ................ ................ 2011-01-25 22:15 +0000 [r303963] Terry Wilson * /, channels/chan_sip.c: Merged revisions 303962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ ................ ................ 2011-01-25 18:56 +0000 [r303861] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 303860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303860 | tilghman | 2011-01-25 12:55:27 -0600 (Tue, 25 Jan 2011) | 12 lines Merged revisions 303858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) | 5 lines Fix "sip show user ", so that it actually shows results, instead of just completing the last entry. (closes issue #16675) Reported by: pj ........ ................ 2011-01-25 17:58 +0000 [r303772] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ 2011-01-25 17:05 +0000 [r303679] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 303678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines Merged revisions 303677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ ................ ................ 2011-01-25 15:52 +0000 [r303638] Matthew Nicholson * main/utils.c: Use unsigned char in comparison for UTF8 check to quiet a compiler warning. 2011-01-24 20:57 +0000 [r303547-303551] Russell Bryant * main/channel.c, main/pbx.c, /, apps/app_meetme.c, main/features.c, include/asterisk/channel.h: Merged revisions 303549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ ................ * contrib/scripts/install_prereq: Add gsm-devel as a package to install on redhat based systems. 2011-01-24 18:59 +0000 [r303509] Matthew Nicholson * res/res_config_curl.c, include/asterisk/utils.h, funcs/func_curl.c, channels/chan_sip.c, tests/test_utils.c, res/res_agi.c, channels/sip/reqresp_parser.c, main/http.c, main/utils.c, funcs/func_uri.c: According to section 19.1.2 of RFC 3261: For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ 2011-01-24 17:21 +0000 [r303468] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 303467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303467 | qwell | 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ ................ ................ 2011-01-22 04:13 +0000 [r303418] Russell Bryant * configure, configure.ac: Revert default compiler change. If someone wishes to do so, it is trivial to set your own default when running the configure script. 2011-01-21 23:11 +0000 [r303288-303376] Jason Parker * channels/chan_dahdi.c, /: Temporarily revert r303288 * channels/chan_dahdi.c, /: Merged revisions 303286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303286 | qwell | 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ ................ ................ 2011-01-21 09:09 +0000 [r303198-303235] Tilghman Lesher * configure, configure.ac: Really use llvm-gcc, when available. * funcs/func_db.c, CHANGES: Add DB_KEYS. Discussion on #asterisk on 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) leifmadsen: you need to know the tree structure already though (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan (02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything (02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script (02:10:13 PM) boch: for example i need to know all entries in the tree (02:10:15 PM) boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) leifmadsen: that's the problem (02:10:22 PM) boch: thank you (02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over. (02:15:35 PM) leifmadsen: database shows everything (02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show ' (02:16:41 PM) leifmadsen: guess no one has found that important enough to program :) (02:16:52 PM) leifmadsen: at that point you should probably just use a relational database... (02:17:10 PM) mateu: i dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does (02:17:24 PM) leifmadsen: no one has programmed it though (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen (02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS() (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family? (02:34:02 PM) leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate through layers of them? (02:34:30 PM) leifmadsen: i.e. family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, yes 2011-01-20 20:35 +0000 [r303154] Richard Mudgett * /, main/ccss.c: Merged revisions 303153 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines Merged revision 303098 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines CC_INTERFACES does not get built correctly with local channels. If local channels are used with CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended to it. * Initialize the buffer to eliminate the prepended garbage. * Filter out the empty interface strings to eliminate the latter. * Added a diagnostic message if the CC_INTERFACES is ever empty. JIRA ABE-2740 JIRA SWP-2848 .......... ................ 2011-01-20 19:58 +0000 [r303108] Shaun Ruffell * /, main/features.c: Merged revisions 303107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303107 | sruffell | 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) | 23 lines Merged revisions 303106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) | 15 lines main/features: Use POLLPRI when waiting for events on parked channels. This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue #18262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 ........ ................ 2011-01-20 17:14 +0000 [r303011] Jeff Peeler * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions 303009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ ................ ................ 2011-01-20 16:12 +0000 [r302922] Russell Bryant * /, apps/app_privacy.c: Merged revisions 302921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines Resolve a compiler warning. ........ ................ 2011-01-20 15:46 +0000 [r302919] Leif Madsen * apps/app_dial.c, /: Merged revisions 302918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines Option L() is milliseconds, not seconds. > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) ........ ................ 2011-01-20 09:07 +0000 [r302879] Tilghman Lesher * configure, configure.ac: On systems which have LLVM, use that compiler. Should result in a massive speed increase. 2011-01-19 23:57 +0000 [r302838] Russell Bryant * main/manager.c, /: Merged revisions 302837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2 lines Only check container count if it exists. ........ 2011-01-19 23:53 +0000 [r302835-302836] Sean Bright * main/config.c: Clarify a source comment about configuration template categories. (closes issue #18578) Reported by: astmiv Patches: asterisk.main.config.2.patch uploaded by astmiv (license 1189) * /, apps/app_voicemail.c: Merged revisions 302834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302834 | seanbright | 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) | 7 lines Support greetingsfolder as documented in voicemail.conf.sample. (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) ........ ................ 2011-01-19 23:33 +0000 [r302832] Paul Belanger * /, contrib/scripts/install_prereq: Merged revisions 302831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan 2011) | 2 lines Add binutils-dev for BETTER_BACKTRACES ........ 2011-01-19 23:07 +0000 [r302786-302790] Russell Bryant * main/manager.c, /: Merged revisions 302789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302789 | russell | 2011-01-19 17:06:46 -0600 (Wed, 19 Jan 2011) | 11 lines Merged revisions 302788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) | 4 lines Turn a noisy verbose message into a debug message. This can drown your console if you're using the AMI over HTTP. ........ ................ * main/manager.c, /: Merged revisions 302785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines Resolve a memory leak with the manager interface is disabled. The intent of this check as it stands in previous versions of Asterisk was to check if there are any active sessions. If there were no sessions, then the function would return immediately and not bother with queueing up the manager event to be processed. Since the conversion of storing sessions in an astobj2 container, this check will always pass. I changed it to go back to checking what was intended. The side effect of this was that if the AMI is disabled, the manager event queue is populated anyway, but the code that runs to clear out the queue never runs. A producer with no consumer is a bad thing. Reported internally by kmorgan. ........ 2011-01-19 21:35 +0000 [r302732] Richard Mudgett * /, main/features.c: Merged revisions 302713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines Merged revisions 302693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ ................ ................ 2011-01-19 21:24 +0000 [r302644-302686] Tilghman Lesher * /, include/asterisk/astdb.h: Merged revisions 302680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600 (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ ................ ................ * /, main/app.c: Merged revisions 302634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302634 | tilghman | 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) | 22 lines Merged revisions 302599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied ........ ................ 2011-01-19 20:15 +0000 [r302601] Jason Parker * /, res/res_fax.c: Merged revisions 302600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) | 1 line Fix typo pointed out on asterisk-users list. ........ 2011-01-19 19:04 +0000 [r302507-302556] Sean Bright * /, main/utils.c: Merged revisions 302555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302555 | seanbright | 2011-01-19 14:03:32 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines Don't call strlen() when we only need to look at the next character or two. (closes issue #18042) Reported by: wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717) ........ ................ * /, main/features.c: Merged revisions 302552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302552 | seanbright | 2011-01-19 13:54:47 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan 2011) | 7 lines Remove an extraneous \r\n at the end of a parking manager events. (closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) ........ ................ * /, res/res_agi.c: Merged revisions 302549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302549 | seanbright | 2011-01-19 13:43:11 -0500 (Wed, 19 Jan 2011) | 17 lines Merged revisions 302548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines Properly handle partial reads from fgets() when handling AGIs. When fgets() failed with EAGAIN, we were continually decrementing the available space left in our buffer, resulting in botched command handling. (closes issue #16032) Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by fnordian (license 110) ........ ................ * /, main/utils.c: Merged revisions 302505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302505 | seanbright | 2011-01-19 12:58:11 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan 2011) | 7 lines Make sure that h_length is set when we short-circuit out of ast_gethostbyname. (closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) ........ ................ 2011-01-19 17:15 +0000 [r302463] Paul Belanger * /, res/res_timing_timerfd.c: Merged revisions 302462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302462 | pabelanger | 2011-01-19 12:09:35 -0500 (Wed, 19 Jan 2011) | 9 lines Merged revisions 302461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed, 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable' error more gracefully. ........ ................ 2011-01-19 15:54 +0000 [r302413-302418] Sean Bright * /, configs/extensions.conf.sample: Merged revisions 302417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302417 | seanbright | 2011-01-19 10:53:20 -0500 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan 2011) | 9 lines Remove references to priorityjumping from the sample extensions.conf. Priority jumping was removed from pbx_config in r68970. (closes issue #18622) Reported by: kshumard Patches: extensions.conf.sample.patch uploaded by kshumard (license 92) ........ ................ * /, channels/chan_sip.c: Merged revisions 302414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) | 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ * channels/chan_local.c, /: Merged revisions 302412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ 2011-01-18 22:06 +0000 [r302319] Richard Mudgett * /, main/features.c: Merged revisions 302318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) | 1 line Use the expanded format type instead of plain int. ........ 2011-01-18 21:44 +0000 [r302315] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 302314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ ................ ................ 2011-01-18 20:40 +0000 [r302270] Jeff Peeler * main/pbx.c, /: Merged revisions 302266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines Merged revisions 302265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip. Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ ........ ................ 2011-01-18 20:21 +0000 [r302268] Russell Bryant * /, main/astobj2.c: Merged revisions 302267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on. AO2_DEBUG is not important and is causing a false compiler warning to be generated on my Ubuntu Natty dev box. ........ 2011-01-18 18:17 +0000 [r302178] Richard Mudgett * /, main/features.c: Merged revisions 302174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302174 | rmudgett | 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) | 102 lines Merged revisions 302173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ ................ ................ 2011-01-17 16:38 +0000 [r302006-302048] Terry Wilson * /, channels/chan_sip.c: Merged revisions 293493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ * /, configs/sip.conf.sample: Merged revisions 302005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines Document "encryption" option in sip.conf.sample ........ 2011-01-14 21:13 +0000 [r301947] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 301946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines Deadlock between dahdi_request() and pri_dchannel() processing an incomming call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. ........ 2011-01-14 20:18 +0000 [r301858] Brett Bryant * channels/chan_multicast_rtp.c, /: Merged revisions 301851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) | 6 lines Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ ........ 2011-01-14 20:07 +0000 [r301850] Andrew Latham * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to function documentation. Fix amatuer type mistake 2011-01-14 19:44 +0000 [r301847] Brett Bryant * channels/chan_multicast_rtp.c, /: Merged revisions 301845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) | 9 lines Fix for a consistent MulticastRTP channel driver crash due to use of unitilized data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ ........ 2011-01-14 19:39 +0000 [r301846] Andrew Latham * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to function documentation. 2011-01-14 17:34 +0000 [r301791] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 301790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman ........ 2011-01-13 17:02 +0000 [r301732] Leif Madsen * /, configs/phoneprov.conf.sample: Merged revisions 301731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines Add static entry for split Polycom 332 firmware. (closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama ........ ................ 2011-01-13 16:27 +0000 [r301729] Paul Belanger * main/pbx.c, CHANGES: Add dialplan variables for asterisk.conf directories Review: https://reviewboard.asterisk.org/r/1075/ 2011-01-12 21:24 +0000 [r301684] Terry Wilson * /, channels/chan_sip.c: Merged revisions 301683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 | twilson | 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) | 9 lines Don't reject all SUBSCRIBE auth requests When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh ........ ................ 2011-01-12 18:52 +0000 [r301596] Matthew Nicholson * main/manager.c, /: Merged revisions 301595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301595 | mnicholson | 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) | 22 lines Merged revisions 301594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't be necessary in session_do, and removed the ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ ................ ................ 2011-01-12 18:12 +0000 [r301505] Jeff Peeler * main/channel.c, /: Merged revisions 301504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ ................ ................ 2011-01-12 16:05 +0000 [r301447] David Vossel * /, main/file.c: Merged revisions 301446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) | 2 lines Removal of unused variables so Asterisk will compile. ........ 2011-01-12 15:59 +0000 [r301445] Stefan Schmidt * Makefile: fix wrong text of rerun menuselect after user interface warning the warning, if no user interface for menuselect warning was found is not right. you have to rerun configure before make menuselect after installing a proper user interface. (closes issue 0018594) Reported by: Dovid 2011-01-12 00:27 +0000 [r301403] Tilghman Lesher * /, main/file.c: Merged revisions 301402 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) | 7 lines Call execl() directly for a better solution for paths with spaces. (closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) ........ 2011-01-11 19:19 +0000 [r301319] Paul Belanger * /, configs/extensions.conf.sample: Merged revisions 301311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301311 | pabelanger | 2011-01-11 14:16:06 -0500 (Tue, 11 Jan 2011) | 9 lines Merged revisions 301310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan 2011) | 2 lines Fix a logic issue when passing context ARG ........ ................ 2011-01-11 18:55 +0000 [r301309] Matthew Nicholson * /, main/utils.c: Merged revisions 301308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301308 | mnicholson | 2011-01-11 12:51:40 -0600 (Tue, 11 Jan 2011) | 18 lines Merged revisions 301307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600 (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ ................ ................ 2011-01-10 22:40 +0000 [r301264] Tilghman Lesher * /, main/strcompat.c: Merged revisions 301263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301263 | tilghman | 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011) | 8 lines Little endian machines were not converted properly. (closes issue #18583) Reported by: jcovert Patches: 20110110__issue18583.diff.txt uploaded by tilghman (license 14) Tested by: jcovert ........ 2011-01-09 21:42 +0000 [r301178-301222] Paul Belanger * /, configure, configure.ac, autoconf/ast_ext_lib.m4: Merged revisions 301221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301221 | pabelanger | 2011-01-09 16:40:34 -0500 (Sun, 09 Jan 2011) | 21 lines Merged revisions 301220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue #18589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ ........ ................ * /, apps/app_verbose.c: Merged revisions 301177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301177 | pabelanger | 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) | 14 lines Merged revisions 301176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) | 7 lines Indicate log level argument for Log() is not optional (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) ........ ................ 2011-01-08 01:13 +0000 [r301135] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 301134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) | 7 lines The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is not dialable. Make a channel name like DAHDI/i3/400-12 dialable when the sequence number is stripped off of the name. ........ 2011-01-07 20:53 +0000 [r301091] Jason Parker * /, apps/app_meetme.c: Merged revisions 301090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines Merged revisions 301089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas ........ ................ 2011-01-07 19:58 +0000 [r301048] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 301047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301047 | jpeeler | 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) | 15 lines Merged revisions 301046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines Fix regression causing forwarding voicemails to not work with file storage. I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 ........ ................ 2011-01-07 18:23 +0000 [r301008] Tilghman Lesher * funcs/func_curl.c: Oops, missed the actual decoding part. (closes issue #18046) Reported by: wdoekes 2011-01-07 17:24 +0000 [r300959] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 300955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300955 | jpeeler | 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) | 21 lines Merged revisions 300951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ ................ ................ 2011-01-07 07:47 +0000 [r300882] Mark Murawki * res/res_config_pgsql.c: Added support for postgres database retry query on disconnection to res_config_pgsql If your postgres connection died suddenly in between res_config_pgsql queries, the next query will fail because the query is executed on a disconnected/disconnecting handle. The query is abandoned and is returned from in error. Now we will reconnect and try again if a query was run on a disconnected connection. (closes issue #18071) 2011-01-06 17:50 +0000 [r300799-300841] Tilghman Lesher * Makefile, funcs/func_curl.c: XML validation * funcs/func_curl.c: Add a hashcompat mode called "legacy", which translates a literal plus sign to a space. (closes issue #18046) Reported by: wdoekes Patches: 20100930__issue18046.diff.txt uploaded by tilghman (license 14) * /, addons/res_config_mysql.c: Merged revisions 300798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r300798 | tilghman | 2011-01-06 00:28:18 -0600 (Thu, 06 Jan 2011) | 8 lines Don't destroy handle not created by use (because the caller will). (closes issue #18526) Reported by: makoto Patches: res-config-mysql-include.patch uploaded by makoto (license 38) Tested by: makoto ........ 2011-01-06 01:41 +0000 [r300761] David Ruggles * Makefile, contrib/scripts/safe_asterisk: update safe_asterisk script change defaults to make a little more sense. Default log location is now asterisk log location and default email notification has been changed to root on the local machine Review: https://reviewboard.asterisk.org/r/1067/ 2011-01-05 21:07 +0000 [r300716] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 300714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300714 | rmudgett | 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) | 21 lines Merged revision 300711 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines A call retrieved from hold may wind up with no audio. If the retrieved call is natively bridged then the call may not have any audio path. The following warning message is given: "Failed to add to conference /: Invalid argument". * Open the media on a B channel when pri_fixup_principle() moves the call from a no_b_channel channel to a real channel. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. .......... ................ 2011-01-05 18:57 +0000 [r300624] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 300623 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300623 | tilghman | 2011-01-05 12:56:12 -0600 (Wed, 05 Jan 2011) | 24 lines Merged revisions 300622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600 (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines Use the sanity check in place of the disconnect/connect cycle. The disconnect/connect cycle has the potential to cause random crashes. (closes issue #18243) Reported by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) Tested by: ks3 ........ ................ ................ 2011-01-05 16:30 +0000 [r300576] Paul Belanger * /, cdr/cdr_sqlite.c: Merged revisions 300575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300575 | pabelanger | 2011-01-05 11:29:19 -0500 (Wed, 05 Jan 2011) | 13 lines Merged revisions 300574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan 2011) | 6 lines Change deprecated message to LOG_WARNING Also removed latter part of message Discussed on #asterisk-dev ........ ................ 2011-01-04 21:54 +0000 [r300434-300522] Leif Madsen * /, channels/chan_sip.c, channels/chan_agent.c, channels/chan_iax2.c, main/xmldoc.c: Merged revisions 300521 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) ........ ................ * configs/users.conf.sample, /: Merged revisions 300433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300433 | lmadsen | 2011-01-04 15:00:55 -0600 (Tue, 04 Jan 2011) | 15 lines Merged revisions 300431 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011) | 7 lines Add some documentation to users.conf.sample. (closes issue #18531) Reported by: lathama Patches: users.conf.sample2.diff uploaded by lathama (license 1028) Tested by: lathama ........ ................ 2011-01-04 21:00 +0000 [r300432] Russell Bryant * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport: Merged revisions 300430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300430 | russell | 2011-01-04 15:00:16 -0600 (Tue, 04 Jan 2011) | 18 lines Merged revisions 300429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300429 | russell | 2011-01-04 14:59:56 -0600 (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) | 4 lines Update the autosupport script from Digium support. (closes AST-395) ........ ................ ................ 2011-01-04 19:45 +0000 [r300385] Leif Madsen * phoneprov/000000000000.cfg, /: Merged revisions 300384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r300384 | lmadsen | 2011-01-04 13:45:22 -0600 (Tue, 04 Jan 2011) | 7 lines Update STAT() to use the comma instead of the pipe. (closes issue #18503) Reported by: cjacobsen Patches: old_separator.diff uploaded by cjacobsen (license 1029) Tested by: lathama ........ 2011-01-04 18:51 +0000 [r300345] Moises Silva * channels/chan_dahdi.c: Update MFC-R2 code to use new DTMF-R2 functionality in OpenR2 (closes issue #18576) 2011-01-04 18:06 +0000 [r300302] Terry Wilson * /, channels/chan_sip.c: Merged revisions 300301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300301 | twilson | 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) | 29 lines Merged revisions 300298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ ................ ................ 2011-01-04 17:04 +0000 [r300215] Jan Kalab * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /: Merged revisions 300214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r300214 | pitel | 2011-01-04 18:01:52 +0100 (Út, 04 led 2011) | 7 lines Memory leaking in calendars ne_request_destroy() was missing in icalendar and exchange calendar modules, causing memory leak. (closes issue #18521) Review: https://reviewboard.asterisk.org/r/1068/ ........ 2011-01-04 16:38 +0000 [r300168-300212] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, UPGRADE.txt, CHANGES, channels/sig_pri.c: Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ * /, main/features.c: Merged revisions 300166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300166 | rmudgett | 2011-01-03 17:14:55 -0600 (Mon, 03 Jan 2011) | 11 lines Merged revisions 300165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) | 4 lines Use correct variable for atxfercallbackretries config option. * Misc formatting changes. ........ ................ 2011-01-03 14:09 +0000 [r300121] David Ruggles * apps/app_externalivr.c: initialize playing_silence in struct initialization playing_silence was not initialized with the struct was initialized, it was being set after the fact which caused problems if something that relied on playing_silence being set was called too quickly (closes issue #18430) Reported by: stevebrandli Patches: externalivr.patch uploaded by thedavidfactor (license 903) Tested by: thedavidfactor, stevebrandli 2011-01-03 13:15 +0000 [r300083] Leif Madsen * /, pbx/pbx_dundi.c: Merged revisions 300082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011) | 11 lines Increase side of mapping response field. I've increased the size of the response field in a DUNDi mapping because of some documentation I'm writing. Previously it was set to AST_MAX_EXTENSION which is only 80 characters, which is far too small when you're using some dialplan functions to craft a response. The example I'm using is: extensions => RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial ........ 2010-12-31 09:29 +0000 [r300044-300045] Tilghman Lesher * cdr/cdr_adaptive_odbc.c, CHANGES, configs/cdr_adaptive_odbc.conf.sample: Support negative filters. (closes issue #17979) Reported by: tilghman Patches: 20100911__for_blitzrage.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen * main/logger.c, CHANGES: Support an alternate configuration file for the 'logger reload' command. (closes issue #17668) Reported by: tilghman Patches: 20100718__logger_reload_altconf__2.diff.txt uploaded by tilghman (license 14) Review: (by lmadsen, russell within comments on issue tracker) 2010-12-29 22:19 +0000 [r300000] Sean Bright * main/asterisk.c: Remove some trailing whitespace and steal revision 300000. 2010-12-29 22:03 +0000 [r299990] Tilghman Lesher * /, main/file.c, apps/app_voicemail.c: Merged revisions 299989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines Quote arguments, just in case there's a space in a pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by me.) ........ 2010-12-29 19:29 +0000 [r299866-299949] Paul Belanger * /, sounds/Makefile: Merged revisions 299948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299948 | pabelanger | 2010-12-29 14:28:36 -0500 (Wed, 29 Dec 2010) | 2 lines Only remove /tmp/astdatadir, not /var/lib/asterisk ........ * Makefile, /, build_tools/make_sample_voicemail, sounds/Makefile: Merged revisions 299907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299907 | pabelanger | 2010-12-29 13:22:23 -0500 (Wed, 29 Dec 2010) | 2 lines Properly quote varibles for MAC OS X ........ * /, apps/app_chanspy.c: Merged revisions 299865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299865 | pabelanger | 2010-12-28 13:53:37 -0500 (Tue, 28 Dec 2010) | 9 lines Merged revisions 299864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) | 2 lines Documentation typo ........ ................ 2010-12-27 21:23 +0000 [r299754-299824] Tilghman Lesher * /, sounds/Makefile: Merged revisions 299820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299820 | tilghman | 2010-12-27 15:23:10 -0600 (Mon, 27 Dec 2010) | 2 lines More space-in-pathname issues. ........ * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged revisions 299794 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299794 | tilghman | 2010-12-27 14:41:04 -0600 (Mon, 27 Dec 2010) | 2 lines Mac OS X spaces-in-pathnames fix. ........ * /, configure, configure.ac: Merged revisions 299752 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299752 | tilghman | 2010-12-26 15:15:58 -0600 (Sun, 26 Dec 2010) | 2 lines Properly quote path on Darwin. ........ 2010-12-25 16:35 +0000 [r299715] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c: Change order of sending TCS and MSD packets Change order of sending Terminal Capability Set and MasterSlave Determination packets, MSD send when TCS exchange procedure is done (we send tcs ack to remote and we have remote tcs ack already or we receive tcs ack from remote and we have send our tcs ack to remote already). Some endpoints can work in this sequence only, i suggest they can't work with both (tcs and msd) exchange procedures simultaneously. Also changed StartH245 facility message sending. It send on incoming calls only due to some endpoints can't proccess properly this facility messages on their incoming calls. (closes issue #18433) Reported by: MrHanMan Patches: tcs-msd-h245-3.patch uploaded by may213 (license 454) Tested by: MrHanMan, may213 2010-12-25 10:08 +0000 [r299584-299627] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 299626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299626 | tilghman | 2010-12-25 04:07:15 -0600 (Sat, 25 Dec 2010) | 19 lines Merged revisions 299625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600 (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ ................ ................ * /, addons/res_config_mysql.c: Merged revisions 299583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299583 | tilghman | 2010-12-24 11:58:30 -0600 (Fri, 24 Dec 2010) | 7 lines Reset 'first' variable after usage. (closes issue #18525) Reported by: makoto Patches: res-config-mysql-update2.patch uploaded by makoto (license 38) ........ 2010-12-23 01:46 +0000 [r299493] Moises Silva * channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue #18438) Reported by: mariner7 Tested by: moy 2010-12-22 20:10 +0000 [r299450] Tilghman Lesher * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest25, pbx/ael/ael-test/ref.ael-vtest17, /, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 299449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299449 | tilghman | 2010-12-22 14:05:02 -0600 (Wed, 22 Dec 2010) | 15 lines Merged revisions 299448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) | 8 lines Resolve warnings by disambiguating the "s" extension as used by chan_dahdi from the "s" extension as used by the AEL macros. (closes issue #18480) Reported by: nivek Patches: 20101215__issue18480__2.diff.txt uploaded by tilghman (license 14) Tested by: nivek ........ ................ 2010-12-22 02:12 +0000 [r299406] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 299405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) | 17 lines Chan_dahdi sends an empty COLP on the bridged channel. Chan_dahdi always inserts a connected party IE when you call from one dahdi channel to another dahdi channel, even if no such information was received on the 2nd channel. This clears the display of many phones. * Removed leftover artifact from before the valid flag was added. * Updated all of the channel's caller id information with the new connected line information instead of just the string parts. (closes issue #18508) Reported by: wimpy Patches: issue18508_trunk.patch uploaded by rmudgett (license 664) Tested by: wimpy, rmudgett ........ 2010-12-21 16:02 +0000 [r299355] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 299353 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299353 | mnicholson | 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) | 30 lines Merged revisions 299242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) | 23 lines Merged revisions 299194,299198,299220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use ast_free() instead of free() ABE-2656 ........ ................ ................ 2010-12-21 00:45 +0000 [r299313] Paul Belanger * configs/cel.conf.sample, /: Merged revisions 299312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299312 | pabelanger | 2010-12-20 19:44:08 -0500 (Mon, 20 Dec 2010) | 8 lines Correct typo with USER_DEFINED event. (closes issue #18461) Reported by: joscas Patches: cel.conf.sample.diff uploaded by lathama (license 1028) Tested by: lathama, joscas ........ 2010-12-20 21:40 +0000 [r299249] Mark Michelson * /, channels/chan_sip.c: Merged revisions 299248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) | 20 lines Fix a couple of CCSS issues. * Make sure to allocate a cc_params structure when creating autopeers. * Use sip_uri_cmp when retrieving SIP CC agents and monitors in case parameters appear in the URI. (closes issue #18504) Reported by: kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches: 18338.diff uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ 2010-12-20 18:18 +0000 [r299142] Tilghman Lesher * /, sample.call: Merged revisions 299138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299138 | tilghman | 2010-12-20 12:17:28 -0600 (Mon, 20 Dec 2010) | 9 lines Merged revisions 299136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 Dec 2010) | 2 lines Documentation fix ........ ................ 2010-12-20 18:03 +0000 [r299135] David Vossel * include/asterisk/astobj2.h, main/astobj2.c: New astobj2 flag for issuing a callback without locking the container. 2010-12-20 17:59 +0000 [r299133-299134] Russell Bryant * channels/chan_misdn.c: Fix chan_misdn build after sched API changes. * addons/chan_ooh323.c, addons/chan_mobile.c: Fix some build errors in addons due to sched API changes. 2010-12-20 17:48 +0000 [r299132] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 299131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299131 | tilghman | 2010-12-20 11:47:10 -0600 (Mon, 20 Dec 2010) | 18 lines Merged revisions 299130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010) | 11 lines If a call was not answered, then the billsec was calculated unusually large. Also, due to a copy and paste error, a request for the answer field would have given the start value, instead. (closes issue #18460) Reported by: joscas Patches: 20101215__issue18460.diff.txt uploaded by tilghman (license 14) Tested by: joscas ........ ................ 2010-12-20 17:15 +0000 [r299091] Russell Bryant * channels/chan_unistim.c, main/udptl.c, res/res_rtp_asterisk.c, include/asterisk.h, main/rtp_engine.c, main/dnsmgr.c, channels/chan_sip.c, main/ccss.c, include/asterisk/channel.h, channels/chan_gtalk.c, tests/test_sched.c, channels/chan_iax2.c, res/res_rtp_multicast.c, main/channel.c, main/cdr.c, channels/chan_jingle.c, channels/chan_skinny.c, channels/sip/include/globals.h, res/res_stun_monitor.c, channels/sip/dialplan_functions.c, channels/chan_h323.c, include/asterisk/sched.h, pbx/pbx_dundi.c, include/asterisk/udptl.h, include/asterisk/rtp_engine.h, main/sched.c, channels/chan_mgcp.c, res/res_calendar.c: Some scheduler API cleanup and improvements. Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ 2010-12-20 16:19 +0000 [r299089] Leif Madsen * /, main/features.c: Merged revisions 299088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299088 | lmadsen | 2010-12-20 10:18:26 -0600 (Mon, 20 Dec 2010) | 13 lines Merged revisions 299087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010) | 5 lines Note that Park() timeout is milliseconds. (closes issue #15758) Reported by: mmurdock Tested by: mmurdock, seanbright ........ ................ 2010-12-20 09:14 +0000 [r299005] Tzafrir Cohen * channels/sig_pri.h, channels/chan_sip.c, main/aoc.c: Typos: recieved => received 2010-12-18 00:08 +0000 [r298819-298961] Tilghman Lesher * utils/refcounter.c, include/asterisk/utils.h, build_tools/cflags-devmode.xml, /, main/Makefile, include/asterisk/autoconfig.h.in, configure.ac, utils/hashtest.c, main/utils.c, main/astobj2.c, utils/conf2ael.c, include/asterisk/logger.h, configure, build_tools/menuselect-deps.in, main/logger.c, utils/hashtest2.c, utils/ael_main.c, makeopts.in, utils/check_expr.c: Merged revisions 298960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. Review: https://reviewboard.asterisk.org/r/1055/ ........ ................ ................ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 298827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298827 | tilghman | 2010-12-17 15:18:18 -0600 (Fri, 17 Dec 2010) | 8 lines -v implies -f, so override with -F. (closes issue #18446) Reported by: lathama Patches: rc.debian.asterisk.diff uploaded by lathama (license 1028) Tested by: lathama ........ * /, configure, configure.ac: Merged revisions 298818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298818 | tilghman | 2010-12-17 15:04:21 -0600 (Fri, 17 Dec 2010) | 15 lines Merged revisions 298817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17 Dec 2010) | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS for SQLite 3, as it's needed on some platforms. (closes issue #18493) Reported by: pprindeville Patches: asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347) Tested by: pprindeville ........ ................ 2010-12-17 17:29 +0000 [r298774] Brad Watkins * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 298773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) | 10 lines Fix parsing of mwi => lines in sip.conf Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. (closes issue #18350) Reported by: gbour Tested by: Marquis, gbour Review: https://reviewboard.asterisk.org/r/1053/ ........ 2010-12-16 23:33 +0000 [r298599-298686] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 298685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298685 | jpeeler | 2010-12-16 17:31:50 -0600 (Thu, 16 Dec 2010) | 16 lines Merged revisions 298684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600 (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines After recording only silence for a voicemail prepending, restore backup files. ........ ................ ................ * /, apps/app_queue.c: Merged revisions 298598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines Merged revisions 298597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ ................ ................ 2010-12-16 09:29 +0000 [r298441-298545] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 298539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) | 8 lines Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (closes issue #18464) Reported by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG (license 20) (plus a few additional lines by tilghman) ........ * res/res_config_odbc.c, /: Merged revisions 298482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298482 | tilghman | 2010-12-16 03:05:28 -0600 (Thu, 16 Dec 2010) | 28 lines Merged revisions 298481 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600 (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010) | 14 lines Only increment the pointer once per loop, otherwise we corrupt the value. (closes issue #18251) Reported by: bcnit Patches: 20101110__issue18251.diff.txt uploaded by tilghman (license 14) Tested by: trev, jthurman, elguero (closes issue #18279) Reported by: zerohalo Patches: 20101109__issue18279.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ ................ * /, funcs/func_dialgroup.c: Merged revisions 298478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298478 | tilghman | 2010-12-16 02:56:13 -0600 (Thu, 16 Dec 2010) | 15 lines Merged revisions 298477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010) | 8 lines Eliminate duplicates from container. (closes issue #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt uploaded by tilghman (license 14) Tested by: bunny ........ ................ * /, cdr/cdr_sqlite.c: Merged revisions 298394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298394 | tilghman | 2010-12-15 18:30:04 -0600 (Wed, 15 Dec 2010) | 22 lines Merged revisions 298393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600 (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010) | 8 lines Unregister before shutting down the connection, to avoid a race. (closes issue #18481) Reported by: pabelanger Patches: 20101215__issue18481.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger ........ ................ ................ 2010-12-13 22:10 +0000 [r298201-298288] Richard Mudgett * channels/sig_pri.c: Post AMI hold events on PRI spans when the remote party HOLD/RETRIEVEs the call. Part of JIRA SWP-2687/ABE-2691. * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 298195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298195 | rmudgett | 2010-12-13 11:11:43 -0600 (Mon, 13 Dec 2010) | 33 lines Merged revisions 298194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered transfers. Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 ........ ................ ................ 2010-12-12 03:58 +0000 [r298137] Jeff Peeler * include/asterisk/utils.h, configure, include/asterisk/autoconfig.h.in, configure.ac, main/logger.c, main/utils.c, main/asterisk.c: Add support for several platforms to obtain the real thread ID. Already had the pthread ID which is not the same. The most obvious enhancement is in the "core show threads" output. As stated in the utils header, if the platform isn't supported -1 is reported (instead of the process ID previously). 2010-12-11 21:47 +0000 [r298100] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c: Correction to work with gatekeeper which don't send GK ID Don't use GK ID if it's not presented in GK replies Extract GK ID not only in GK confirm but in GK register confirm also (closes issue #18401) Reported by: MrHanMan Patches: no-gkid-2.patch uploaded by may213 (license 454) Tested by: may213, MrHanMan 2010-12-10 16:53 +0000 [r298055] Matthew Nicholson * /, res/res_fax.c: Merged revisions 298054 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298054 | mnicholson | 2010-12-10 10:52:11 -0600 (Fri, 10 Dec 2010) | 2 lines Prevent a memcpy overlap in GENERIC_FAX_EXEC_SET_VARS ........ 2010-12-10 16:28 +0000 [r298052] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/netsock.c: Merged revisions 298051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298051 | tilghman | 2010-12-10 10:26:46 -0600 (Fri, 10 Dec 2010) | 18 lines Merged revisions 298050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010) | 11 lines Portability issue on OpenSolaris. Also detect the required structure element, because OpenSolaris defines SIOCGIFHWADDR, but without support for IP sockets. (closes issue #18442) Reported by: ranjtech Patches: 20101209__issue18442.diff.txt uploaded by tilghman (license 14) Tested by: ranjtech ........ ................ 2010-12-09 22:19 +0000 [r297972] Terry Wilson * /, channels/chan_sip.c: Merged revisions 297965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297965 | twilson | 2010-12-09 16:18:19 -0600 (Thu, 09 Dec 2010) | 28 lines Merged revisions 297960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297960 | twilson | 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ ................ ................ 2010-12-09 21:33 +0000 [r297958] David Vossel * /, channels/chan_gtalk.c: Merged revisions 297957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09 Dec 2010) | 11 lines Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (closes issue #18412) Reported by: nevermind_quack Patches: fix uploaded by dvossel (license 671) ........ 2010-12-09 21:26 +0000 [r297956] Terry Wilson * /, main/features.c: Merged revisions 297952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297952 | twilson | 2010-12-09 14:48:44 -0600 (Thu, 09 Dec 2010) | 10 lines Don't crash after Set(CDR(userfield)=...) in ast_bridge_call Instead of setting peer->cdr = NULL, set it to not post. (closes issue #18415) Reported by: macbrody Patches: patch-18415 uploaded by jsolares (license 1167) Tested by: jsolares, twilson ........ 2010-12-08 18:08 +0000 [r297910] Tilghman Lesher * /, configs/extensions.conf.sample: Merged revisions 297909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297909 | tilghman | 2010-12-08 12:06:04 -0600 (Wed, 08 Dec 2010) | 11 lines Merged revisions 297908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010) | 4 lines Use inheritance to get correct results for SIPFROMDOMAIN. (from an internal Digium discussion) ........ ................ 2010-12-07 23:00 +0000 [r297826] Jeff Peeler * main/channel.c, /: Merged revisions 297825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600 (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ ................ ................ 2010-12-07 22:54 +0000 [r297734-297822] Tilghman Lesher * Makefile, contrib/init.d/org.asterisk.asterisk.plist, utils/muted.c, /, contrib/init.d/org.asterisk.muted.plist (added): Merged revisions 297821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297821 | tilghman | 2010-12-07 16:51:05 -0600 (Tue, 07 Dec 2010) | 18 lines Merged revisions 297819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600 (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) | 4 lines Use non-deprecated APIs for CoreAudio Review: https://reviewboard.asterisk.org/r/1040/ ........ ................ ................ * /, apps/app_followme.c: Merged revisions 297733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297733 | tilghman | 2010-12-06 18:29:26 -0600 (Mon, 06 Dec 2010) | 22 lines Merged revisions 297713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ ................ ................ 2010-12-06 22:10 +0000 [r297608] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 297607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297607 | jpeeler | 2010-12-06 16:06:37 -0600 (Mon, 06 Dec 2010) | 25 lines Merged revisions 297605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ ................ ................ 2010-12-03 17:42 +0000 [r297536] Sean Bright * /, channels/chan_console.c: Merged revisions 297535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297535 | seanbright | 2010-12-03 12:41:30 -0500 (Fri, 03 Dec 2010) | 9 lines Merged revisions 297534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri, 03 Dec 2010) | 3 lines The CLI command should not contain s, these are for descriptions. ........ ................ 2010-12-03 15:32 +0000 [r297496] Matthew Nicholson * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions 297157,297486,297495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec 2010) | 2 lines Changed some NOTICE and WARNING messages to DEBUG messages. ........ r297486 | mnicholson | 2010-12-02 15:30:47 -0600 (Thu, 02 Dec 2010) | 6 lines Add support for reserving a fax session before answering the channel. Note: this change breaks ABI compatibility. FAX-217 ........ r297495 | mnicholson | 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines Print a DEBUG message instead of a WARNING message when the selected fax tech does not support reserving sessions. Answer the channel before quering it for t.38 support. This is necessary for the query to work properly over local channels. ........ 2010-12-02 20:11 +0000 [r297407] Paul Belanger * Makefile, /: Merged revisions 297406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297406 | pabelanger | 2010-12-02 15:09:29 -0500 (Thu, 02 Dec 2010) | 21 lines Merged revisions 297405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500 (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec 2010) | 7 lines Resolve compile error under FreeBSD We now set _ASTCFLAGS+=-march=i686 for i386 processors, still allowing ASTCFLAGS to override the setting. Review: https://reviewboard.asterisk.org/r/1043/ ........ ................ ................ 2010-12-02 18:28 +0000 [r297356] Terry Wilson * /, main/abstract_jb.c: Merged revisions 297312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297312 | twilson | 2010-12-02 12:13:49 -0600 (Thu, 02 Dec 2010) | 28 lines Merged revisions 297311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297311 | twilson | 2010-12-02 12:07:39 -0600 (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) | 12 lines Initialize offset for adaptive jitter buffer When the adaptive jitter buffer is enabled in sip.conf, the first frame placed in the jitter buffer fails with something like: jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466, threshold 1000, new offset 215886466 This happens because the offset is not initialized before calling jb_put(). This patch modifies jb_put_first_adaptive() to set the offset to the frame's timestamp. Review: https://reviewboard.asterisk.org/r/1041/ ........ ................ ................ 2010-12-02 13:20 +0000 [r297248] Russell Bryant * /, apps/app_meetme.c: Merged revisions 297245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines Merged revisions 297229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ ................ ................ 2010-12-01 17:53 +0000 [r297076] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 297075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ ................ ................ 2010-12-01 17:03 +0000 [r296952-296993] Tilghman Lesher * /, include/asterisk/frame.h: Merged revisions 296992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296992 | tilghman | 2010-12-01 11:01:56 -0600 (Wed, 01 Dec 2010) | 19 lines Merged revisions 296991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600 (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) | 5 lines Clarify documentation on how we store codec preference lists. (closes issue #18397) Reported by: birgita ........ ................ ................ * /, channels/chan_iax2.c: Merged revisions 296951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296951 | tilghman | 2010-11-30 19:46:32 -0600 (Tue, 30 Nov 2010) | 9 lines Merged revisions 296950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30 Nov 2010) | 2 lines Missed initializations caused startup errors on Mac OS X (and possibly others, too). ........ ................ 2010-12-01 00:28 +0000 [r296871] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 296870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296870 | jpeeler | 2010-11-30 18:28:16 -0600 (Tue, 30 Nov 2010) | 18 lines Merged revisions 296869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ ................ ................ 2010-11-30 22:32 +0000 [r296788-296826] Tilghman Lesher * include/asterisk/frame.h: Add a comment on why the reserved bit is reserved. Came up when reviewing discussion on the CODEC PREFS IE in IAX2. * /, apps/app_meetme.c: Merged revisions 296787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) | 2 lines DOC: Conference number can be omitted; if omitted, all users in a meetme are listed. ........ 2010-11-30 09:49 +0000 [r296752] Stefan Schmidt * include/asterisk.h, main/pbx.c, main/asterisk.c: move devices from hints into an ao2_container by splitting up devices from hints into an own ao2_container the callback to get these devices for statechange handling is faster. with this changes the length of a device used in a hint isnt longer restricted to 80 characters. Tests showed that calling handle_statechange is 40 times faster if no hints are used and 25 times faster if there are any hints. (closes issue #17928) Reported by: mdu113 Tested by: schmidts Review: https://reviewboard.asterisk.org/r/1003/ 2010-11-29 23:07 +0000 [r296674] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 296673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296673 | pabelanger | 2010-11-29 18:05:45 -0500 (Mon, 29 Nov 2010) | 19 lines Merged revisions 296671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500 (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov 2010) | 5 lines Make sure nothing else is needed before destroying the scheduler. (closes issue #18398) Reported by: pabelanger ........ ................ ................ 2010-11-29 21:31 +0000 [r296630] Russell Bryant * /, channels/chan_sip.c: Merged revisions 296628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) | 6 lines Complete some error handling in transmit_publish() in chan_sip.c. This error handling block caught my eye. It was missing a couple of things, but it should be safe now. Thanks to mmichelson for the quick peer review on IRC. ........ 2010-11-29 20:54 +0000 [r296585] Richard Mudgett * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 296582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296582 | rmudgett | 2010-11-29 14:46:03 -0600 (Mon, 29 Nov 2010) | 24 lines Merged revision 296575 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon, 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY redirecting number and notification code, SETUP redirecting number) is also sent in PTMP/TE mode. It should only apply in PTMP/NT mode. The call setup proceeds but the network (Deutsche Telekom) reacts with ugly ISDN STATUS messages. Also don't send the redirecting number ie when PTP is also sending the DivertingLegInformation2 facility. The redirecting number ie is redundant and the network (Deutsche Telekom) complains about it. Patches: abe_2651_v4.patch uploaded by rmudgett (license 664) JIRA ABE-2651 JIRA SWP-2537 .......... ................ 2010-11-29 07:30 +0000 [r296535] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/asterisk.c: Merged revisions 296534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296534 | tilghman | 2010-11-29 01:28:44 -0600 (Mon, 29 Nov 2010) | 20 lines Merged revisions 296533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) | 13 lines I love standards. There are so many to choose from. Except when there isn't one. Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (closes issue #18384) Reported by: bjm Patches: cred-diffs uploaded by bjm (license 473) 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14) 20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, bjm ........ ................ 2010-11-27 10:41 +0000 [r296430-296468] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 296467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296467 | tilghman | 2010-11-27 04:40:22 -0600 (Sat, 27 Nov 2010) | 12 lines Merged revisions 296466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) | 5 lines 18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). (closes issue #18369) Reported by: tnakonz ........ ................ * include/asterisk.h, /: Merged revisions 296429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296429 | tilghman | 2010-11-27 03:58:57 -0600 (Sat, 27 Nov 2010) | 5 lines Also don't build DEBUG_FD_LEAKS when STANDALONE2 is defined. (closes issue #18385) Reported by: cmaj ........ 2010-11-26 22:02 +0000 [r296393] Olle Johansson * /, main/say.c: Merged revisions 296391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296391 | oej | 2010-11-26 22:37:21 +0100 (Fre, 26 Nov 2010) | 24 lines Merged revisions 296351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre, 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 lines Fix bugs in saying numbers using the Swedish language syntax (closes issue #18355) Reported by: oej Patch by: oej Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break. Review: https://reviewboard.asterisk.org/r/1033/ ........ ................ ................ 2010-11-26 18:31 +0000 [r296353-296355] Brad Watkins * /, res/res_jabber.c: Merged revisions 296354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296354 | marquis | 2010-11-26 13:31:17 -0500 (Fri, 26 Nov 2010) | 12 lines Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (closes issue #18272) Reported by: klaus3000 Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, Marquis Review: https://reviewboard.asterisk.org/r/1030/ ........ * /, channels/chan_sip.c: Merged revisions 296352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296352 | marquis | 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) | 12 lines Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot. (closes issue #18342) Reported by: nivek Patches: issue0018342p1.patch uploaded by nivek (license 636) Tested by: nivek Review: https://reviewboard.asterisk.org/r/1029/ ........ 2010-11-24 23:46 +0000 [r296249] Andrew Parisio * apps/app_meetme.c, CHANGES: Meetme use voicemail greet for join/leave announce Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command. Review: https://reviewboard.asterisk.org/r/1009/ (closes issue #18297) Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded by parisioa (license 1153) 2010-11-24 23:30 +0000 [r296235] Russell Bryant * main/channel.c, /: Merged revisions 296230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ ................ ................ 2010-11-24 22:52 +0000 [r296168] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions 296167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines Merged revisions 296166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ ................ ................ 2010-11-24 20:24 +0000 [r296034-296085] Russell Bryant * main/channel.c, /: Merged revisions 296084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ ................ ................ * apps/app_dial.c, main/channel.c, /: Merged revisions 296002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ ................ 2010-11-23 10:34 +0000 [r295950] Olle Johansson * /, main/say.c: Merged revisions 295949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295949 | oej | 2010-11-23 11:30:05 +0100 (Tis, 23 Nov 2010) | 21 lines Merged revisions 295907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis, 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 lines Fix support of saynumber(1,n) in the Swedish language (closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ ........ ................ ................ 2010-11-22 20:05 +0000 [r295870] Sean Bright * configs/chan_dahdi.conf.sample, /: Merged revisions 295869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295869 | seanbright | 2010-11-22 15:03:49 -0500 (Mon, 22 Nov 2010) | 9 lines Merged revisions 295868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov 2010) | 2 lines Change some documentation to suggest dahdi_monitor instead of ztmonitor. ........ ................ 2010-11-22 19:42 +0000 [r295867] Richard Mudgett * main/channel.c, main/pbx.c, /, apps/app_macro.c, include/asterisk/channel.h, include/asterisk/frame.h: Merged revisions 295866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ ................ 2010-11-22 18:43 +0000 [r295789] Erin Spiceland * res/res_agi.c: Revert to the previous behavior of AGI command WAIT FOR DIGIT, since the behavior of the command with this patch is almost exactly like that of GET DATA. 2010-11-20 03:13 +0000 [r295748] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions 295747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295747 | rmudgett | 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) | 13 lines One way audio before answering call waiting call on analog port. * Analog call waiting Caller ID spills could get stuck resulting in one way audio until the waiting call is answered. This only happens on the second (and later) call waiting call if the active call is not the first call. * The CLI/AMI "dahdi show channel" command could report the wrong channel information. Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer in sync. ........ 2010-11-20 00:52 +0000 [r295712] Russell Bryant * include/asterisk/event.h, /, main/event.c: Merged revisions 295711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines Merged revisions 295710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines Fix cache of device state changes for multiple servers. This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ ........ ................ 2010-11-19 22:15 +0000 [r295674] Terry Wilson * /, channels/chan_sip.c: Merged revisions 295673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295673 | twilson | 2010-11-19 14:06:10 -0800 (Fri, 19 Nov 2010) | 22 lines Merged revisions 295672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295672 | twilson | 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ ................ ................ 2010-11-19 21:42 +0000 [r295671] Brett Bryant * /, apps/app_queue.c: Merged revisions 295670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) | 8 lines Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ ........ 2010-11-19 19:32 +0000 [r295554] Erin Spiceland * res/res_agi.c: Add extra functionality to AGI command WAIT FOR DIGIT. Add the ability to play a sound file, listen for more than just one digit, specify escape characters. Backwards compatible (to work with only timeout specified). (closes issue #15531) Reported by: diLLec Patches: asterisk-res_agi-203638-patched.patch uploaded by diLLec (license 839) Tested by: diLLec, espiceland 2010-11-19 16:49 +0000 [r295517] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions 295516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295516 | rmudgett | 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) | 13 lines Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed initial value of struct analog_pvt.use_callerid. It may get forced on depending upon other config options. * Call analog_dnd() instead of manual inlined code. * Removed unused struct analog_pvt.usedistinctiveringdetection. * Removed the struct analog_pvt.unknown_alarm flag. It was really the struct analog_pvt.inalarm flag. * Use ast_debug() instead of ast_log(LOG_DEBUG). * Rename several function's index variable to idx. * Some formatting tweaks. ........ 2010-11-18 20:31 +0000 [r295478] Leif Madsen * configs/sip_notify.conf.sample, /: Merged revisions 295477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295477 | lmadsen | 2010-11-18 14:30:35 -0600 (Thu, 18 Nov 2010) | 6 lines 'sip notify clear-mwi' needs terminating CRLF. (closes issue #18275) Reported by: klaus3000 Patches: fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65) ........ 2010-11-18 18:08 +0000 [r295364-295442] Paul Belanger * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions 295441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295441 | pabelanger | 2010-11-18 13:02:12 -0500 (Thu, 18 Nov 2010) | 11 lines Merged revisions 295440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov 2010) | 4 lines Fix compiler warnings when using openssl-dev 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........ ................ * /, contrib/scripts/install_prereq: Merged revisions 295404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295404 | pabelanger | 2010-11-18 00:12:05 -0500 (Thu, 18 Nov 2010) | 2 lines Add RedHat specific dependencies ........ * /, configs/res_curl.conf.sample: Merged revisions 295361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295361 | pabelanger | 2010-11-17 09:09:38 -0500 (Wed, 17 Nov 2010) | 2 lines Uncomment settings under [global], to surpress warning when loading Asterisk. ........ 2010-11-16 23:04 +0000 [r295283] Richard Mudgett * main/channel.c, /: Merged revisions 295282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600 (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ ................ ................ 2010-11-16 22:41 +0000 [r295125-295279] Russell Bryant * /, build_tools/prep_tarball: Merged revisions 295278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295278 | russell | 2010-11-16 16:41:11 -0600 (Tue, 16 Nov 2010) | 2 lines Check for pdftotext and give a useful error if not found. ........ * /, build_tools/prep_tarball: Merged revisions 295201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295201 | russell | 2010-11-16 15:46:18 -0600 (Tue, 16 Nov 2010) | 2 lines Remove intentional typo I had added when testing the check. oops. ........ * /, build_tools/prep_tarball: Merged revisions 295164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295164 | russell | 2010-11-16 14:50:03 -0600 (Tue, 16 Nov 2010) | 2 lines Check for wikiexport.py in PATH and give a useful error message if not found. ........ * main/app.c: Remove a trailing space. (testing something with bamboo ...) 2010-11-15 19:11 +0000 [r294990-295079] Tilghman Lesher * tests/test_expr.c (added), /: Merged revisions 295078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295078 | tilghman | 2010-11-15 12:30:13 -0600 (Mon, 15 Nov 2010) | 16 lines Merged revisions 295062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600 (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) | 2 lines Create test verifying results of expression parser ........ ................ ................ * funcs/func_curl.c, /: Merged revisions 294989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294989 | tilghman | 2010-11-15 01:44:38 -0600 (Mon, 15 Nov 2010) | 15 lines Merged revisions 294988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) | 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ 2010-11-12 21:15 +0000 [r294907-294912] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 294911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294911 | jpeeler | 2010-11-12 15:14:43 -0600 (Fri, 12 Nov 2010) | 11 lines Merged revisions 294910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) | 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ ................ * /, apps/app_voicemail.c: Merged revisions 294905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294905 | jpeeler | 2010-11-12 14:52:06 -0600 (Fri, 12 Nov 2010) | 30 lines Merged revisions 294904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ ................ 2010-11-12 02:46 +0000 [r294824] Richard Mudgett * channels/sig_pri.h, /, channels/sig_pri.c: Merged revisions 294823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294823 | rmudgett | 2010-11-11 20:45:22 -0600 (Thu, 11 Nov 2010) | 25 lines Merged revisions 294822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ ................ 2010-11-11 22:18 +0000 [r294741-294749] Russell Bryant * /, doc/CCSS_architecture.pdf (removed): Merged revisions 294745 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) | 6 lines Remove CCSS architecture PDF. It has been moved to: https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture ........ * doc/CODING-GUIDELINES (removed), doc/ss7.txt (removed), /, doc/backtrace.txt (removed), doc/India-CID.txt (removed), doc/digium-mib.txt (removed), doc/followme.txt (removed), doc/building_queues.txt (removed), doc/timing.txt (removed), doc/advice_of_charge.txt (removed), doc/unistim.txt (removed), doc/video_console.txt (removed), doc/macroexclusive.txt (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt (added), doc/callfiles.txt (removed), build_tools/prep_tarball, doc/codec-64bit.txt (removed), doc/externalivr.txt (removed), doc/video.txt (removed), doc/jingle.txt (removed), doc/modules.txt (removed), doc/manager_1_1.txt (removed), doc/PEERING (removed), doc/snmp.txt (removed), doc/siptls.txt (removed), doc/HOWTO_collect_debug_information.txt (removed), doc/ldap.txt (removed), doc/sip-retransmit.txt (removed), doc/distributed_devstate.txt (removed), doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed), doc/queue.txt (removed), doc/jabber.txt (removed), doc/chan_sip-perf-testing.txt (removed), doc/asterisk-mib.txt (removed), Makefile, doc/database_transactions.txt (removed), doc/smdi.txt (removed), doc/janitor-projects.txt (removed), doc/hoard.txt (removed), doc/res_config_sqlite.txt (removed), doc/osp.txt (removed), doc/speechrec.txt (removed), doc/sms.txt (removed), doc/distributed_devstate-XMPP.txt (removed), doc/valgrind.txt (removed), doc/realtimetext.txt (removed), doc/cli.txt (removed), doc/rtp-packetization.txt (removed), doc/datastores.txt (removed): Merged revisions 294740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines Remove most of the contents of the doc dir in favor of the wiki content. This merge does the following things: * Removes most of the contents from the doc/ directory in favor of the wiki - http://wiki.asterisk.org/ * Updates the build_tools/prep_tarball script to know how to export the contents of the wiki in both PDF and plain text formats so that the documentation is still included in Asterisk release tarballs. ........ 2010-11-11 22:01 +0000 [r294735] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 294734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294734 | jpeeler | 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) | 32 lines Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ ................ 2010-11-10 23:27 +0000 [r294570-294606] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 294605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294605 | tilghman | 2010-11-10 17:26:39 -0600 (Wed, 10 Nov 2010) | 2 lines Fixing the Mac OS X build (bamboo warning) ........ * pbx/pbx_spool.c, /: Merged revisions 294569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294569 | tilghman | 2010-11-10 17:13:37 -0600 (Wed, 10 Nov 2010) | 8 lines Properly queue files with inotify(7). (closes issue #18089) Reported by: abelbeck Patches: 20101021__issue18089.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-11-10 14:15 +0000 [r294502-294536] Russell Bryant * /, res/ais/clm.c, res/ais/evt.c, UPGRADE-1.8.txt: Merged revisions 294535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) | 5 lines Tweak a couple of CLI commands back to their original form. The "module" in this case is two parts, so there are two words before the verb of the CLI command. ........ * /, main/devicestate.c: Merged revisions 294501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294501 | russell | 2010-11-10 06:46:27 -0600 (Wed, 10 Nov 2010) | 14 lines Merged revisions 294500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) | 7 lines Improve a debug message to be more readable and consistent. (closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) ........ ................ 2010-11-09 22:52 +0000 [r294467] Richard Mudgett * main/channel.c, /: Merged revisions 294466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) | 1 line Allow ast_do_masquerade() failure to be reported again. ........ 2010-11-09 20:35 +0000 [r294431] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 294430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294430 | tilghman | 2010-11-09 14:33:05 -0600 (Tue, 09 Nov 2010) | 15 lines Merged revisions 294429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) | 8 lines Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ 2010-11-09 17:00 +0000 [r294351] Richard Mudgett * main/channel.c, channels/chan_misdn.c, channels/sig_analog.c, /, include/asterisk/channel.h, channels/sig_pri.c: Merged revisions 294349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 ........ 2010-11-08 22:33 +0000 [r294279-294314] Jeff Peeler * /, res/res_timing_timerfd.c: Merged revisions 294313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294313 | jpeeler | 2010-11-08 16:32:13 -0600 (Mon, 08 Nov 2010) | 9 lines Merged revisions 294312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08 Nov 2010) | 1 line add missing unlock not present in 294277 ........ ................ * main/channel.c, /, res/res_timing_timerfd.c, include/asterisk/timing.h, main/timing.c: Merged revisions 294278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ ................ 2010-11-08 21:04 +0000 [r294244] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 294243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294243 | mnicholson | 2010-11-08 14:56:30 -0600 (Mon, 08 Nov 2010) | 15 lines Merged revisions 294242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) | 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ ................ 2010-11-08 19:59 +0000 [r294208] Terry Wilson * /, configs/calendar.conf.sample, res/res_calendar.c: Merged revisions 294207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294207 | twilson | 2010-11-08 13:56:10 -0600 (Mon, 08 Nov 2010) | 2 lines Set a default waittime, and make sure to convert it to milliseconds ........ 2010-11-08 17:19 +0000 [r294127] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ 2010-11-05 22:17 +0000 [r294086] Brett Bryant * /, channels/chan_sip.c: Merged revisions 294084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294084 | bbryant | 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010) | 9 lines Fixed deadlock avoidance issues while locking channel when adding the Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ ........ 2010-11-05 21:56 +0000 [r294083] David Vossel * channels/chan_sip.c: Perform proper handling of forked outbound INVITE requests. RFC3261 section 12 about dialog creation says an INVITE transaction results in an established dialog once it receives the 200 OK response. It is possible to receive multiple differing 200 OK responses for a single outbound INVITE Request, and this should result in establishing multiple dialogs. This patch allows for all differing 200 OK responses to an INVITE request to establish a separate dialog, but only the first dialog is kept. All other resulting dialogs from the initial request are immediately ACKed and then immediately terminated with a BYE request. Review: https://reviewboard.asterisk.org/r/946/ 2010-11-05 16:07 +0000 [r294048-294050] Terry Wilson * contrib/scripts/ast_tls_cert, /: Merged revisions 294049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294049 | twilson | 2010-11-05 09:05:50 -0700 (Fri, 05 Nov 2010) | 2 lines Corret spelling and example ........ * contrib/scripts/ast_tls_cert, /: Merged revisions 294047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294047 | twilson | 2010-11-05 08:36:20 -0700 (Fri, 05 Nov 2010) | 2 lines Tell people to use the correct common name for clients as well ........ 2010-11-05 15:26 +0000 [r294046] David Vossel * /, channels/chan_sip.c: Merged revisions 293924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293924 | dvossel | 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010) | 4 lines Fixes ringback tone on sip semi-attended transfer. ABE-2168 ........ 2010-11-05 00:08 +0000 [r293971] Shaun Ruffell * /, codecs/codec_dahdi.c: Merged revisions 293970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293970 | sruffell | 2010-11-04 19:07:11 -0500 (Thu, 04 Nov 2010) | 32 lines Merged revisions 293969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically commit 9034) added the capability for the wctc4xxp to return more than a single packet of data in response to a read. However, when decoding packets, codec_dahdi was still assuming that the default number of samples was in each read. In other words, each packet your provider sent you, regardless of size, would result in 20 ms of decoded data (30 ms if decoding G723). If your provider was sending 60 ms packets then codec_dahdi would end up stripping 40 ms of data from each transcoded frame resulting in "choppy" audio. This would only affect systems where G729 packets are arriving in sizes greater than 20ms or G723 packets arriving in sizes greater than 30ms. DAHDI-744. ........ ................ ................ 2010-11-04 13:29 +0000 [r293888] Paul Belanger * /, channels/chan_sip.c: Merged revisions 293887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293887 | pabelanger | 2010-11-04 09:27:54 -0400 (Thu, 04 Nov 2010) | 8 lines Do not output port in IPaddress for AMI sippeers. (closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn ........ 2010-11-03 18:43 +0000 [r293809] Terry Wilson * main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 293803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ 2010-11-03 18:38 +0000 [r293808] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 293807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293807 | rmudgett | 2010-11-03 13:35:19 -0500 (Wed, 03 Nov 2010) | 34 lines Merged revisions 293806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ ................ 2010-11-02 23:10 +0000 [r293725] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 293724 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293724 | jpeeler | 2010-11-02 18:09:06 -0500 (Tue, 02 Nov 2010) | 22 lines Merged revisions 293723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ ................ 2010-11-02 21:31 +0000 [r293649] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 293648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293648 | rmudgett | 2010-11-02 16:29:25 -0500 (Tue, 02 Nov 2010) | 20 lines Merged revisions 293647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '' on channel ((), line )". ........ ................ ................ 2010-11-02 20:47 +0000 [r293578-293612] Paul Belanger * main/manager.c, /: Merged revisions 293611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293611 | pabelanger | 2010-11-02 16:45:09 -0400 (Tue, 02 Nov 2010) | 2 lines If manager and tls are disabled, do not display TCP/TLS Bindaddress. ........ * configs/gtalk.conf.sample, UPGRADE.txt, channels/chan_gtalk.c, CHANGES: New CLI command 'gtalk show settings'. Review: https://reviewboard.asterisk.org/r/984/ 2010-11-02 14:43 +0000 [r293577] Mark Michelson * CHANGES: Add to the CHANGES file that the HTTP server supports IPv6 addressing. 2010-11-01 17:32 +0000 [r293531] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions 293530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293530 | rmudgett | 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) | 10 lines Analog 3-way call would not connect all parties if one was using sig_pri. Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. ........ 2010-11-01 16:11 +0000 [r293497] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 293496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293496 | pabelanger | 2010-11-01 12:09:05 -0400 (Mon, 01 Nov 2010) | 13 lines Use ast_sockaddr_from_sin function not memcpy This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 ........ 2010-10-30 01:55 +0000 [r293342-293419] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 293418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293418 | rmudgett | 2010-10-29 20:53:29 -0500 (Fri, 29 Oct 2010) | 16 lines Merged revisions 293417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line Remove some more code that serves no purpose. ........ ................ ................ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 293341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293341 | rmudgett | 2010-10-29 19:46:41 -0500 (Fri, 29 Oct 2010) | 16 lines Merged revisions 293340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line Remove some code that serves no purpose. ........ ................ ................ 2010-10-29 21:50 +0000 [r293306] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 293305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) | 9 lines Modify sip_setoption to not complain about unknown options. This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc ........ 2010-10-29 20:46 +0000 [r293273] Mark Michelson * main/http.c, UPGRADE.txt, configs/http.conf.sample: Enable IPv6 for the built-in HTTP server. Review: https://reviewboard.asterisk.org/r/986 2010-10-28 20:01 +0000 [r293198] Tilghman Lesher * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c, res/ael/ael.tab.h, main/ast_expr2.c: Merged revisions 293197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293197 | tilghman | 2010-10-28 15:00:06 -0500 (Thu, 28 Oct 2010) | 33 lines Merged revisions 293195-293196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ r293196 | tilghman | 2010-10-28 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ ................ 2010-10-28 16:11 +0000 [r293160] Jeff Peeler * /, funcs/func_strings.c: Merged revisions 293159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293159 | jpeeler | 2010-10-28 11:11:08 -0500 (Thu, 28 Oct 2010) | 18 lines Merged revisions 293158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ ................ 2010-10-26 18:54 +0000 [r293120] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 293119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293119 | jpeeler | 2010-10-26 13:49:08 -0500 (Tue, 26 Oct 2010) | 43 lines Merged revisions 293118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ ................ 2010-10-26 16:33 +0000 [r293047-293082] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 293081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293081 | rmudgett | 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010) | 1 line No need to define the struct if there are no users. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Merged revisions 293046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) | 4 lines Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. ........ 2010-10-25 21:16 +0000 [r292915-292970] Tilghman Lesher * /, channels/sig_pri.c: Merged revisions 292969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) | 2 lines Several more defines that need to be altered for compiling against an older version of libpri ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Merged revisions 292906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) | 4 lines Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. ........ 2010-10-25 19:11 +0000 [r292869] David Vossel * channels/chan_local.c, /: Merged revisions 292868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292868 | dvossel | 2010-10-25 14:07:50 -0500 (Mon, 25 Oct 2010) | 39 lines Merged revisions 292867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ ................ 2010-10-22 22:40 +0000 [r292808-292826] Terry Wilson * contrib/scripts/ast_tls_cert, /: Merged revisions 292825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292825 | twilson | 2010-10-22 15:35:29 -0700 (Fri, 22 Oct 2010) | 4 lines Don't create directories without at least o+x Also, making files that you are going to modify read-only is dumb. ........ * contrib/scripts/ast_tls_cert, /: Merged revisions 292794 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292794 | twilson | 2010-10-22 15:18:36 -0700 (Fri, 22 Oct 2010) | 2 lines Make files readable only by the owner ........ 2010-10-22 21:29 +0000 [r292788] Leif Madsen * /, channels/chan_sip.c, configs/res_ldap.conf.sample, contrib/scripts/asterisk.ldif: Merged revisions 292787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292787 | lmadsen | 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) | 21 lines Merged revisions 292786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner ........ ................ 2010-10-22 17:16 +0000 [r292743] Terry Wilson * contrib/scripts/ast_tls_cert (added), /: Merged revisions 292740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) | 45 lines Add TLS cert helper script This script is useful for quickly generating self-signed CA, server, and client certificates for use with Asterisk. It is still recommended to obtain certificates from a recognized Certificate Authority and to develop an understanding how SSL certificates work. Real security is hard work. OPTIONS: -h Show this message -m Type of cert "client" or "server". Defaults to server. -f Config filename (openssl config file format) -c CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed) -k CA key filename -C Common name (cert field) For a server cert, this should be the same address that clients attempt to connect to. Usually this will be the Fully Qualified Domain Name, but might be the IP of the server. For a CA or client cert, it is merely informational. Make sure your certs have unique common names. -O Org name (cert field) An informational string (company name) -o Output filename base (defaults to asterisk) -d Output directory (defaults to the current directory) Example: To create a CA and a server (pbx.mycompany.com) cert with output in /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp This will create a CA cert and key as well as asterisk.pem and the the two files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will require you to import the ca.crt file as a trusted cert. To create a client cert using the CA cert created by the example above: ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user This will create client.crt/key/pem in /tmp. Use this if your device supports a client certificate. Make sure that you have the ca.crt file set up as a tlscafile in the necessary Asterisk configs. Make backups of all .key files in case you need them later. ........ 2010-10-22 17:10 +0000 [r292742] Mark Michelson * /, tests/test_event.c: Merged revisions 292741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct 2010) | 12 lines Prevent multiple runs of event_sub_test from producing false failure results. The array of test subscriptions was declared "static," meaning that the data.count field would retain its value between runs of the test. After the first test run, this would result in false reports of test failures. I chose to just remove the "static" keyword from the structure since it's not a huge deal to construct this structure during each run of the test. Another alternative would have been to zero out the data.count fields of each test subscription instead. ........ 2010-10-22 15:47 +0000 [r292705] Richard Mudgett * main/channel.c, channels/chan_misdn.c, /, channels/sig_pri.c: Merged revisions 292704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ ........ 2010-10-21 22:11 +0000 [r292668] Tilghman Lesher * /, channels/misdn/ie.c: Merged revisions 292667 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292667 | tilghman | 2010-10-21 17:09:25 -0500 (Thu, 21 Oct 2010) | 2 lines Compile correctly on Linux (asterisk/localtime.h depends upon asterisk/autoconfig.h loading first). ........ 2010-10-21 18:23 +0000 [r292630] Paul Belanger * /, contrib/init.d/rc.suse.asterisk: Merged revisions 292628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292628 | pabelanger | 2010-10-21 14:13:18 -0400 (Thu, 21 Oct 2010) | 5 lines Fix typo in SUSE init script. Reported by: Dave Cotton on asterisk-users list. ........ 2010-10-21 16:46 +0000 [r292597] David Vossel * main/manager.c, /: Merged revisions 292595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) | 14 lines Fixes recursive lock problem in manager.c It is possible for a AMI session to freeze because of invalid use of recursive locks during the EVENT processing. This patch removes the unnecessary locks. (closes issue #18167) Reported by: sustav Patches: manager_locking_v1.diff uploaded by dvossel (license 671) Tested by: sustav ........ 2010-10-21 13:17 +0000 [r292559] Leif Madsen * /, configs/res_ldap.conf.sample: Merged revisions 292557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292557 | lmadsen | 2010-10-21 08:12:19 -0500 (Thu, 21 Oct 2010) | 14 lines Merged revisions 292556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010) | 6 lines Change res_ldap.sample.conf to match the schema. (closes issue #17376) Reported by: jcovert Patches: res_ldap.conf.sample.patch uploaded by jcovert (license 551) ........ ................ 2010-10-21 11:38 +0000 [r292524] Russell Bryant * /, res/res_config_ldap.c: Merged revisions 292523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292523 | russell | 2010-10-21 06:36:47 -0500 (Thu, 21 Oct 2010) | 4 lines Add var=value to log message on update failure, and add newline. ... just for you, Leif. ........ 2010-10-21 01:03 +0000 [r292490] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 292489 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) | 7 lines Send CONNECT_ACKNOWLEDGE for CIS calls too. The originator of the Q.SIG call completion signaling link was not changed to the active state when the CONNECT message came in. The T309 processing would immediately kill the signaling link because it was not in the active state. ........ 2010-10-21 00:23 +0000 [r292414-292443] Paul Belanger * /, apps/app_voicemail.c: Merged revisions 292436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292436 | pabelanger | 2010-10-20 20:21:59 -0400 (Wed, 20 Oct 2010) | 8 lines Application not properly unregister in voicemail (closes issue #18128) Reported by: junky Patches: vm_unregister.diff uploaded by junky (license 177) Tested by: pabelanger, lmadsen ........ * apps/app_dial.c, /: Merged revisions 292413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292413 | pabelanger | 2010-10-20 20:07:17 -0400 (Wed, 20 Oct 2010) | 24 lines Merged revisions 292412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ ................ ................ 2010-10-20 00:41 +0000 [r292377] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 292376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19 Oct 2010) | 5 lines Oops. This module uses the generic timer and no longer uses DAHDI. This causes a problem with the Solaris and other system builds that have gcc 4.1 (where optional_api is non-optional). ........ 2010-10-19 22:19 +0000 [r292345] Paul Belanger * /, contrib/scripts/install_prereq: Merged revisions 292343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292343 | pabelanger | 2010-10-19 18:14:23 -0400 (Tue, 19 Oct 2010) | 2 lines Add resample and imap_tk dependencies. ........ 2010-10-19 19:35 +0000 [r292310] Terry Wilson * /, channels/chan_sip.c, res/res_srtp.c: Merged revisions 292309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines Add sip show peer info about crypto and remove dated comment This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko ........ 2010-10-18 22:14 +0000 [r292231] Leif Madsen * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 292225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292225 | lmadsen | 2010-10-18 16:51:23 -0500 (Mon, 18 Oct 2010) | 24 lines Merged revisions 292224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500 (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) | 9 lines Add support for the new English (Australian Accent) sound files. (closes issue #17426) Reported by: camsown Patches: core-sounds-en_AU.txt uploaded by camsown (license 1050) add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested by: camsown, lmadsen, jtodd, qwell ........ ................ ................ 2010-10-18 21:56 +0000 [r292228] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 292227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292227 | jpeeler | 2010-10-18 16:55:46 -0500 (Mon, 18 Oct 2010) | 25 lines Merged revisions 292226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ ................ ................ 2010-10-18 19:52 +0000 [r292189] Russell Bryant * main/netsock2.c, /: Merged revisions 292188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010) | 9 lines Resolve some compiler errors in ast_sockaddr_is_any(). These errors came up once this function was used from within netsock2.c. The errors were like the following: netsock2.c:393: error: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules The usage of a union here avoids this problem. ........ 2010-10-18 19:16 +0000 [r292156] David Vossel * main/netsock2.c, /: Merged revisions 292155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292155 | dvossel | 2010-10-18 14:16:00 -0500 (Mon, 18 Oct 2010) | 2 lines Fixes build error for systems not supporting IPV6_TCLASS. ........ 2010-10-18 17:18 +0000 [r292124] Matthew Nicholson * /, addons/chan_mobile.c: Merged revisions 292122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292122 | mnicholson | 2010-10-18 12:15:24 -0500 (Mon, 18 Oct 2010) | 5 lines Fix the cmgr parser. (closes issue 0018152) Reported by: menschentier ........ 2010-10-18 16:03 +0000 [r292086] David Vossel * main/netsock2.c, /: Merged revisions 292085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292085 | dvossel | 2010-10-18 11:02:17 -0500 (Mon, 18 Oct 2010) | 7 lines Fixes qos settings for sockets bound to any IPv6 or IPv4 address. (closes issue #18099) Reported by: jamesnet Patches: issues_18099_v3.diff uploaded by dvossel (license 671 ........ 2010-10-18 15:33 +0000 [r292084] Jeff Peeler * pbx/pbx_spool.c, /: Merged revisions 292083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292083 | jpeeler | 2010-10-18 10:32:40 -0500 (Mon, 18 Oct 2010) | 4 lines Disable use of inotify for call file handling as it is not working properly. (related to #18089) ........ 2010-10-16 11:51 +0000 [r292052] Tzafrir Cohen * /, configs/musiconhold.conf.sample, res/res_musiconhold.c: Merged revisions 292050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292050 | tzafrir | 2010-10-16 12:47:00 +0200 (ש', 16 אוק 2010) | 22 lines Merged revisions 292049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) | 15 lines Base directory for MOH should be ASTDATADIR If the directive 'directory' is relative, make it relative to the datadir, rather than to the varlibdir. In the sample configuration it is relative ('moh'). This has no effect unless you have actively set the datadir explicitly (at build time or at run time). (closes issue #16906) Patches: moh_datadir uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/974/ ........ ................ 2010-10-15 21:49 +0000 [r292017] Terry Wilson * /, res/res_srtp.c: Merged revisions 292016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) | 5 lines Ref/unref res_srtp when we create/destroy a session This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp tries to unload before chan_sip does. Thanks, Russell! ........ 2010-10-15 20:12 +0000 [r291943] David Vossel * /, channels/chan_sip.c: Merged revisions 291942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291942 | dvossel | 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010) | 8 lines Fixes peer's host port information being lost on sip reload. (closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen ........ 2010-10-15 19:53 +0000 [r291941] Paul Belanger * configs/gtalk.conf.sample, /: Merged revisions 291940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291940 | pabelanger | 2010-10-15 15:50:22 -0400 (Fri, 15 Oct 2010) | 16 lines Merged revisions 291939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400 (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct 2010) | 2 lines Clean up formatting. ........ ................ ................ 2010-10-15 16:54 +0000 [r291906] Terry Wilson * /, res/res_jabber.c: Merged revisions 291905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291905 | twilson | 2010-10-15 09:39:58 -0700 (Fri, 15 Oct 2010) | 14 lines Merged revisions 291904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines Don't crash or deadlock on module unload We can't hold the lock while pthread_join is called since aji_log_hook will attempt to lock from the other therad. We reorder the pthread_join and ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is running, causing a crash. ........ ................ 2010-10-14 22:10 +0000 [r291828-291830] David Vossel * main/netsock2.c, /: Merged revisions 291829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291829 | dvossel | 2010-10-14 17:09:32 -0500 (Thu, 14 Oct 2010) | 8 lines Set TCLASS field of IPv6 header when sip qos options are set. (closes issue #18099) Reported by: jamesnet Patches: issues_18099_v2.diff uploaded by dvossel (license 671) Tested by: dvossel, jamesnet ........ * /, channels/chan_gtalk.c: Merged revisions 291827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14 Oct 2010) | 18 lines Safer xml parsing, treat all clients the same, and better local candidate selection. The gtalk channel driver was doing several unsafe operations in regards to how it parsed incoming XML messages. I have cleaned that code up so it should be much safer now. We now treat all clients types the same. We have no reason to distinguish between GMAIL and GOOGLE VOICE clients anymore because they all work the same way. I also modified how the local ip is found. If no bindaddress is provided in the config file, we attempt to determine the local ip we would use to connect to google.com. If that fails, then we fall back to the ast_find_ourip() function as a last resort. Using the new method makes it much less likely that we would ever advertise a local RTP candidate as a loopback address. ........ 2010-10-14 18:46 +0000 [r291792] Jeff Peeler * /, main/stdtime/localtime.c: Merged revisions 291791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291791 | jpeeler | 2010-10-14 13:45:02 -0500 (Thu, 14 Oct 2010) | 10 lines Add missing ifdefs for test framework and new locale code. (closes issue #18137) Reported by: ovi Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes (license 717) 18137_localelist_warning.patch uploaded by wdoekes (license 717) Tested by: ovi ........ 2010-10-14 15:21 +0000 [r291760] Paul Belanger * channels/chan_jingle.c, include/asterisk/acl.h, /, channels/chan_sip.c, channels/chan_h323.c, main/acl.c, channels/chan_gtalk.c: Merged revisions 291758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines Add the ability for ast_find_ourip to return IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ ........ 2010-10-14 12:10 +0000 [r291726] Russell Bryant * /, doc/tex/secure-calls.tex: Merged revisions 291725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 Oct 2010) | 2 lines Fix a typo - s/seucre/secure/ ........ 2010-10-13 23:52 +0000 [r291658] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions 291656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291656 | rmudgett | 2010-10-13 18:45:11 -0500 (Wed, 13 Oct 2010) | 34 lines Merged revisions 291655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ ................ ................ 2010-10-13 23:47 +0000 [r291657] Terry Wilson * main/channel.c, /: Merged revisions 291581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291581 | twilson | 2010-10-13 16:01:56 -0700 (Wed, 13 Oct 2010) | 35 lines Merged revisions 291580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines Don't ignore frames that have been queued when softhangup'd When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ ........ ................ ................ 2010-10-13 22:47 +0000 [r291579] David Vossel * /, channels/chan_gtalk.c: Merged revisions 291578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291578 | dvossel | 2010-10-13 17:46:34 -0500 (Wed, 13 Oct 2010) | 4 lines More fixup for chan_gtalk. This patch makes the xml parsing safer. ........ 2010-10-13 22:34 +0000 [r291576] Terry Wilson * Makefile, /, static-http/mantest.html (added): Merged revisions 291575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291575 | twilson | 2010-10-13 15:24:44 -0700 (Wed, 13 Oct 2010) | 8 lines Add a simple AMI client web page This patch uses the XML docs to parse all of the available AMI commands and allows you to enter the command name and be presented with a form with the available fields. You can then rapidly tab through the fields and submit the command and view the response. It is much faster/easier than having to use telnet for testing purposes. ........ 2010-10-13 20:24 +0000 [r291470-291542] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 291541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291541 | rmudgett | 2010-10-13 15:21:02 -0500 (Wed, 13 Oct 2010) | 26 lines The chan_dahdi faxdetect option only works for the first FAX call. The chan_dahdi faxdetect option only works for the first call. After that the option no longer works. The struct dahdi_pvt.callprogress member is the encoded user config setting for the callprogress and faxdetect config options. Changing this value alters the configuration for all following calls until the chan_dahdi.conf file is reloaded. * Fixed the chan_dahdi ast_channel_setoption callback to not change the users faxdetect config setting except for the current call. * Fixed the chan_dahdi ast_channel_queryoption callback to read the active DSP setting of the faxdetect option. * Made actually disable the active faxdetect DSP setting for the current call on the analog port. my_handle_dtmfup() is used for normal analog ports. dahdi_handle_dtmfup() is the legacy code and is no longer used unless in a radio mode. (closes issue #18116) Reported by: seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/972/ ........ * channels/chan_misdn.c, /: Merged revisions 291507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291507 | rmudgett | 2010-10-13 14:01:48 -0500 (Wed, 13 Oct 2010) | 18 lines Merged revision 291504 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the ast_channel. Must get the ast_channel lock before proceeding with release_chan() and release_chan_early() to hold off ast_hangup() from destroying the ast_channel. Missed this change for -r291468. JIRA ABE-2598 JIRA SWP-2317 .......... ................ * channels/chan_misdn.c, /: Merged revisions 291469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291469 | rmudgett | 2010-10-13 13:10:21 -0500 (Wed, 13 Oct 2010) | 23 lines Merge revision 291468 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE --> RELEASE_COMPLETE * Add lock protection around channel list for find/add/delete operations. * Protect misdn_hangup() from release_chan() and vise versa using the release_lock. JIRA ABE-2598 JIRA SWP-2317 .......... ................ 2010-10-13 15:51 +0000 [r291395] Russell Bryant * /, channels/chan_sip.c: Merged revisions 291394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291394 | russell | 2010-10-13 10:46:39 -0500 (Wed, 13 Oct 2010) | 20 lines Merged revisions 291393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ ................ ................ 2010-10-13 08:58 +0000 [r291361] Stefan Schmidt * apps/app_macro.c: Report what extension called a failed macro Add the extension and context of the calling channel to the log output if a macro could not be found. (closes issue #18112) Reported by: prado Patches: app_macro-info.diff uploaded by prado (license 510) Tested by: schmidts 2010-10-12 17:21 +0000 [r291287] Leif Madsen * /, configs/phoneprov.conf.sample: Merged revisions 291284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291284 | lmadsen | 2010-10-12 12:20:43 -0500 (Tue, 12 Oct 2010) | 15 lines Merged revisions 291280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) | 7 lines Add undocumented variables to phoneprov.conf.sample (closes issue #18107) Reported by: lathama Patches: phoneprov.conf.sample.diff uploaded by lathama (license 1028) ........ ................ 2010-10-12 17:07 +0000 [r291266] Tilghman Lesher * /, main/acl.c: Merged revisions 291265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291265 | tilghman | 2010-10-12 12:06:23 -0500 (Tue, 12 Oct 2010) | 16 lines Merged revisions 291264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500 (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines Oops, incorrect range (although unallocated at ARIN) ........ ................ ................ 2010-10-12 16:08 +0000 [r291231] Leif Madsen * /, configs/manager.conf.sample: Merged revisions 291230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291230 | lmadsen | 2010-10-12 11:08:04 -0500 (Tue, 12 Oct 2010) | 10 lines Merged revisions 291229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010) | 2 lines Add documention that mentions options are defined but not used. (Issue #18101) ........ ................ 2010-10-12 16:00 +0000 [r291193-291228] David Vossel * main/manager.c, /: Merged revisions 291227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) | 16 lines Fixes manager.c crash. This issue was caused by improper use of the mansession lock and manession_session lock. These two structures are confusing to begin with so I'm not surprised this occurred. I fixed this by consistently making sure we use each of these locks only to protect the data in the corresponding structure. We had mismatched usage of these locks which resulted in no mutual exclusivity occurring at all. (closes issue #17994) Reported by: vrban Patches: mansession_locking_fix.diff uploaded by dvossel (license 671) Tested by: vrban ........ * /, CHANGES: Merged revisions 291194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010) | 2 lines Update CHANGES to reflect new gtalk.conf options. ........ * configs/gtalk.conf.sample, /, res/res_stun_monitor.c, channels/chan_gtalk.c, include/asterisk/stun.h: Merged revisions 291192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines Gtalk enhancements and general code cleanup. This patch includes several chan_gtalk enhancements. Two new gtalk.conf options have been added, externip and stunadd. Setting externip allows us to manually specify what the external IP address is outside of a NAT environment. Setting the stunaddr option to a valid stun server allows for that external ip to be retrieved via a STUN server automatically. This external IP is then advertised during call setup as a possible candidate. I have also attempted to clean up chan_gtalk's code so it meets our coding guidelines. During this cleanup I noticed several things that need to be done in the code and made a TODO section at the top of the file. ........ 2010-10-11 19:07 +0000 [r291076-291115] Richard Mudgett * channels/chan_sip.c: Add todo comment about handle_incoming() calling assumption. * /, channels/chan_sip.c: Merged revisions 291112-291113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291112 | rmudgett | 2010-10-11 13:48:15 -0500 (Mon, 11 Oct 2010) | 20 lines Merged revisions 291110-291111 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ ................ r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit from handle_request_do() consistent. ................ ................ r291113 | rmudgett | 2010-10-11 13:51:13 -0500 (Mon, 11 Oct 2010) | 1 line Move declaration closer to where now used. ................ * /, main/cli.c: Merged revisions 291075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291075 | rmudgett | 2010-10-11 11:42:54 -0500 (Mon, 11 Oct 2010) | 22 lines Merged revisions 291073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) | 15 lines Fixed infinite loop in verbose/debug message output. Setting the module/filename specific message level and then changing it resulted in the linked list being looped on itself. Traversing this linked list is an infinite loop if what you are looking for is not in the list. Also plugged some CLI parsing holes in the associated CLI command: * Removing a nonexistent module from the list actually added it with a level of zero. * Setting the non-module specific level to zero is now equivalent to setting it to "off" as documented. ........ ................ 2010-10-11 03:20 +0000 [r291039] Tilghman Lesher * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged revisions 291038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291038 | tilghman | 2010-10-09 18:25:37 -0500 (Sat, 09 Oct 2010) | 2 lines Add missing option to set calls to be logged in GMT/UTC. ........ 2010-10-09 14:04 +0000 [r291006] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling options per user and peer. Added options for faststart/h.245 tunneling per user/peer, properly handle these and global options, correction of handling fs/tunneling fields in signalling responses (closes issue #17972) Reported by: salecha Patches: fs-tunnel-per-point-3.patch uploaded by may213 (license 454) Tested by: may213, salecha 2010-10-08 20:45 +0000 [r290974] David Vossel * /, channels/chan_gtalk.c: Merged revisions 290973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 Oct 2010) | 12 lines Make outbound Google Voice calls. This patch allows for outbound Google Voice calls to be dialed from Asterisk using chan_gtalk. Below is an example dialstring. exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In this example, 'asterisk' is the jabber.conf profile configured to connect to your gmail account. In order to receive Google Voice calls make sure to enable 'allowguest=yes' in gtalk.conf. ........ 2010-10-08 16:27 +0000 [r290939] Erin Spiceland * addons/app_mysql.c, configs/res_config_mysql.conf.sample, /, addons/res_config_mysql.c: Add option to res_config_mysql and app_mysql to specify a character set that MySQL should use. (closes issue 17948) Reported by qmax. 2010-10-08 03:00 +0000 [r290865] Jeff Peeler * /, main/asterisk.c: Merged revisions 290864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290864 | jpeeler | 2010-10-07 21:56:24 -0500 (Thu, 07 Oct 2010) | 23 lines Merged revisions 290863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500 (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed at control console. A recent change was made to avoid a race condition on shutdown which only called the end functions from the console thread. However, when pressing Ctrl-C the quit handler is called from the signal handler thread. (closes issue #17698) Reported by: jmls ........ ................ ................ 2010-10-07 22:39 +0000 [r290830-290831] David Vossel * /, channels/chan_gtalk.c: Merged revisions 290829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 Oct 2010) | 6 lines Add Philippe Sultan to chan_gtalk author list. Philippe has made some notable contributions to the gtalk channel driver. His name deserves to be listed amoung the authors of that file. Thanks Philippe! ........ * /, channels/chan_gtalk.c: Merged revisions 290828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07 Oct 2010) | 5 lines Outbound gtalk calls now work correctly. There was a problem with how the candidates were being built on an outbound call. This patch fixes that. ........ 2010-10-07 20:59 +0000 [r290753] Jason Parker * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_ext_lib.m4: Merged revisions 290752 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290752 | qwell | 2010-10-07 15:58:47 -0500 (Thu, 07 Oct 2010) | 23 lines Merged revisions 290751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290751 | qwell | 2010-10-07 15:57:14 -0500 (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines Allow PRI to build properly when using --with-pri. Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) ........ ................ ................ 2010-10-07 11:12 +0000 [r290714] Russell Bryant * main/pbx.c, /: Merged revisions 290713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290713 | russell | 2010-10-07 13:00:52 +0200 (Thu, 07 Oct 2010) | 11 lines Merged revisions 290712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010) | 4 lines Don't crash when Set() is called without a value. Review: https://reviewboard.asterisk.org/r/949/ ........ ................ 2010-10-06 21:23 +0000 [r290649-290677] David Vossel * /, channels/chan_gtalk.c: Merged revisions 290674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06 Oct 2010) | 4 lines Fixes commented out code to use #if 0 instead. Thanks to rmudgett for catching this! ........ * /, channels/chan_gtalk.c: Merged revisions 290648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) | 12 lines Fixes gtalk outbound DTMF to work properly. Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered this after investigating a packet capture from the gmail client. Instead of performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue with negotiating RTP payload type 106 for the telephony-event and then sending DTMF as payload 101. This has been resolved by always negotiating 101 as the payload type like we do everywhere else. With this patch, incoming google voice calls forwarded to Asterisk via gtalk work. ........ 2010-10-06 18:56 +0000 [r290615] Richard Mudgett * apps/app_dial.c, /: Merged revisions 290614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290614 | rmudgett | 2010-10-06 13:50:37 -0500 (Wed, 06 Oct 2010) | 12 lines Merged revision 290613 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) | 5 lines Eliminate a redundant test for AST_CONTROL_REDIRECTING. Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running the redirecting interception macro if it is defined. .......... ................ 2010-10-06 13:50 +0000 [r290577] Tilghman Lesher * /, main/file.c: Merged revisions 290576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290576 | tilghman | 2010-10-06 08:49:19 -0500 (Wed, 06 Oct 2010) | 15 lines Merged revisions 290575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) | 8 lines Allow streaming audio from a pipe. (closes issue #18001) Reported by: jamicque Patches: 20100926__issue18001.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ ................ 2010-10-06 04:47 +0000 [r290543] Terry Wilson * res/res_rtp_asterisk.c, /: Merged revisions 290542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05 Oct 2010) | 6 lines Don't try to send RTP when remote_address is null It is possible for ast_rtp_stop() to be called which will clear the remote address and cause the sendto to fail and spam warnings. Don't send in this case. ........ 2010-10-05 22:23 +0000 [r290480-290509] David Vossel * /, channels/chan_iax2.c: Merged revisions 290506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290506 | dvossel | 2010-10-05 17:23:00 -0500 (Tue, 05 Oct 2010) | 2 lines Fixes uninitialized memory problem in 'iax2 set debug peer' option. ........ * /, include/asterisk/jabber.h, include/asterisk/jingle.h, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 290479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) | 6 lines Fixes chan_gtalk to work with gmail client This patch was written by Philippe Sultan (phsultan). Thanks for keeping this up to date! ........ 2010-10-05 20:24 +0000 [r290414] Tilghman Lesher * /, res/res_jabber.c: Merged revisions 290408 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290408 | tilghman | 2010-10-05 15:23:33 -0500 (Tue, 05 Oct 2010) | 22 lines Merged revisions 290396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500 (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) | 8 lines Fix a crash by ensuring that we don't alter memory after it's freed. (closes issue #17387) Reported by: jmls Patches: 20100726__issue17387.diff.txt uploaded by tilghman (license 14) Tested by: jmls ........ ................ ................ 2010-10-05 20:10 +0000 [r290377-290379] David Vossel * /, channels/chan_iax2.c: Merged revisions 290378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290378 | dvossel | 2010-10-05 15:09:06 -0500 (Tue, 05 Oct 2010) | 11 lines Resolves dnsmgr memory corruption in chan_iax2. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: afried, russell, dvossel Review: https://reviewboard.asterisk.org/r/965/ ........ * /, apps/app_directed_pickup.c: Merged revisions 290376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290376 | dvossel | 2010-10-05 14:56:29 -0500 (Tue, 05 Oct 2010) | 16 lines Merged revisions 290375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) | 10 lines Fixes PickupChan() not working with full channel name. (closes issue #18011) Reported by: schern Patches: app_directed_pickup.c.2.patch uploaded by schern (license 995) app_directed_pickup.c.trunk.patch uploaded by schern (license 995) Tested by: schern, dvossel ........ ................ 2010-10-05 14:17 +0000 [r290067-290291] Tilghman Lesher * /, configure, configure.ac: Merged revisions 290289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290289 | tilghman | 2010-10-05 09:15:46 -0500 (Tue, 05 Oct 2010) | 2 lines Restore run directory for OS X, as well as standardizing some other paths to Mac OS X. ........ * res/ael/pval.c, main/pbx.c, pbx/ael/ael-test/ref.ael-vtest17, /, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 290255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290255 | tilghman | 2010-10-04 18:23:11 -0500 (Mon, 04 Oct 2010) | 18 lines Merged revisions 290254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored. Also change the AEL parser to not generate dashes within extensions, as those dashes would be ignored. Update the AEL tests to match this behavior. (closes issue #17366) Reported by: murf Patches: 20100727__issue17366.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ * /, configure, configure.ac: Merged revisions 290209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290209 | tilghman | 2010-10-04 15:23:13 -0500 (Mon, 04 Oct 2010) | 16 lines Merged revisions 290201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500 (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........ ................ ................ * /, configure, configure.ac: Merged revisions 290102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290102 | tilghman | 2010-10-03 16:08:45 -0500 (Sun, 03 Oct 2010) | 16 lines Merged revisions 290101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500 (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 Oct 2010) | 2 lines Automatically re-run configure test for menuselect, when the relevant makeopts settings change. ........ ................ ................ * pbx/pbx_spool.c, /: Merged revisions 290066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290066 | tilghman | 2010-10-03 15:02:29 -0500 (Sun, 03 Oct 2010) | 8 lines Get notification only when file is closed, not when created. (closes issue #17924) Reported by: mkeuter Patches: asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946) Tested by: abelbeck ........ 2010-10-02 18:06 +0000 [r290027] Kevin P. Fleming * contrib/scripts/get_mp3_source.sh, /: Merged revisions 290026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290026 | kpfleming | 2010-10-02 12:57:13 -0500 (Sat, 02 Oct 2010) | 6 lines Allow users to pass additional arguments to the Subversion command that obtains the MP-3 source code. (reported on IRC by jmls) ........ 2010-10-02 08:58 +0000 [r289952] Olle Johansson * main/manager.c, /: Merged revisions 289951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289951 | oej | 2010-10-02 10:56:08 +0200 (Lör, 02 Okt 2010) | 16 lines Merged revisions 289950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör, 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines Add documentation for undocumented option to AMI action originate ........ ................ ................ 2010-10-02 04:54 +0000 [r289876] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 289875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289875 | tilghman | 2010-10-01 23:46:43 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen ........ ................ ................ 2010-10-02 02:46 +0000 [r289841] Jeff Peeler * res/res_rtp_asterisk.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 289840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ ................ 2010-10-01 17:22 +0000 [r289732] Paul Belanger * /, configs/jabber.conf.sample, res/res_jabber.c: Merged revisions 289718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289718 | pabelanger | 2010-10-01 13:19:49 -0400 (Fri, 01 Oct 2010) | 20 lines Merged revisions 289704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400 (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct 2010) | 6 lines Disable debugging by default and reformat .config file. Review: https://reviewboard.asterisk.org/r/929/ ........ ................ ................ 2010-10-01 16:23 +0000 [r289702] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 289701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289701 | jpeeler | 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) | 28 lines Merged revisions 289700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ ................ ................ 2010-10-01 10:04 +0000 [r289623] Stefan Schmidt * channels/chan_sip.c: don't iterate through all dialogs to find and delete old subscribes On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. Review: https://reviewboard.asterisk.org/r/901/ 2010-09-30 20:40 +0000 [r289588] Tilghman Lesher * /, tests/test_time.c, funcs/func_env.c, tests/test_utils.c, res/res_agi.c, include/asterisk/localtime.h, main/stdtime/localtime.c: Merged revisions 289543,289581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) | 2 lines More Solaris compatibility fixes ........ r289581 | tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines Solaris fixes. ........ 2010-09-30 19:54 +0000 [r289555] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 289554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289554 | mnicholson | 2010-09-30 14:53:10 -0500 (Thu, 30 Sep 2010) | 11 lines Merged revisions 289553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines Properly handle channel allocation failures duing invites with replaces. ABE-2588 ........ ................ 2010-09-30 19:35 +0000 [r289552] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 289549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289549 | rmudgett | 2010-09-30 14:28:36 -0500 (Thu, 30 Sep 2010) | 17 lines Merged revision 289547 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, 30 Sep 2010) | 10 lines In chan_misdn, the DivertingLegInformation2 DivertingNr is garbage when the number is restricted. The same thing happens with DivertingLegInformation1 DivertedTo number. The misdn_PresentedNumberUnscreened_extract() extracted the Unscreened PartyNumber field unconditionally. It now checks the presented number unscreened type to see if the PartyNumber was even present. JIRA ABE-2595 .......... ................ 2010-09-30 15:40 +0000 [r289427] Russell Bryant * /, apps/app_sms.c: Merged revisions 289426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289426 | russell | 2010-09-30 10:39:45 -0500 (Thu, 30 Sep 2010) | 22 lines Merged revisions 289425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289425 | russell | 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines Fix a crash in app_sms. Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 ........ ................ ................ 2010-09-29 21:19 +0000 [r289354] Jason Parker * main/channel.c, /, main/features.c: Merged revisions 289340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289340 | qwell | 2010-09-29 16:12:43 -0500 (Wed, 29 Sep 2010) | 22 lines Merged revisions 289339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289339 | qwell | 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines Allow a manager originate to succeed on forwarded devices. The timeout to wait for an answer was being set to 0 when a device forwarded to another extension. We don't always need the timeout set like this, so make it an optional parameter, and don't use it in this case. ABE-2544 ........ ................ ................ 2010-09-29 20:29 +0000 [r289337] Leif Madsen * /, configs/res_ldap.conf.sample: Merged revisions 289336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289336 | lmadsen | 2010-09-29 15:27:25 -0500 (Wed, 29 Sep 2010) | 9 lines Merged revisions 289334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010) | 1 line Update sample documentation to note md5secret requirements. ........ ................ 2010-09-29 20:24 +0000 [r289335] Russell Bryant * /, res/res_config_ldap.c: Merged revisions 289333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289333 | russell | 2010-09-29 15:20:23 -0500 (Wed, 29 Sep 2010) | 11 lines Merged revisions 289332 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29 Sep 2010) | 4 lines Don't completely ignore md5secret from LDAP if the value does not begin with {md5}. This fixes a problem that lmadsen ran in to where md5secret was not working for him. ........ ................ 2010-09-29 17:54 +0000 [r289269-289301] Matthew Nicholson * /, configs/res_fax.conf.sample: Merged revisions 289300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289300 | mnicholson | 2010-09-29 12:53:54 -0500 (Wed, 29 Sep 2010) | 2 lines Add 'ecm' to the sample fax config file ........ * main/channel.c, /: Merged revisions 289268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289268 | mnicholson | 2010-09-29 12:08:20 -0500 (Wed, 29 Sep 2010) | 5 lines Update the CDR record when ast_channel_set_caller_event() is called (related to issue #17569) Reported by: tbelder ........ 2010-09-29 16:17 +0000 [r289254] Richard Mudgett * main/channel.c, /: Merged revisions 289253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289253 | rmudgett | 2010-09-29 11:16:47 -0500 (Wed, 29 Sep 2010) | 1 line Make development error message indicate which channel. ........ 2010-09-29 15:07 +0000 [r289180] Matthew Nicholson * main/channel.c, /: Merged revisions 289179 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289179 | mnicholson | 2010-09-29 10:04:56 -0500 (Wed, 29 Sep 2010) | 22 lines Merged revisions 289178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500 (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines Set the caller id on CDRs when it is set on the parent channel. (closes issue #17569) Reported by: tbelder Patches: 17569.diff uploaded by tbelder (license 618) Tested by: tbelder ........ ................ ................ 2010-09-28 18:24 +0000 [r289131] Brett Bryant * main/channel.c, /: Merged revisions 289099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289099 | bbryant | 2010-09-28 14:18:02 -0400 (Tue, 28 Sep 2010) | 28 lines Merged revisions 289095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400 (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines Fixes an issue with the Newchannel AMI event during the Masquerading process. Fixes an issue with the Newchannel AMI event during the Masquerading process, where no Newchannel AMI event was generated for the psuedo channel used during the masquerading process. (closes issue #17987) Reported by: RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish (license 1122) Tested by: RadicAlish Review: https://reviewboard.asterisk.org/r/937/ ........ ................ ................ 2010-09-28 18:20 +0000 [r289112] Tilghman Lesher * Makefile, /, tests/test_time.c, configure, include/asterisk/autoconfig.h.in, include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c, configure.ac, makeopts.in, apps/app_voicemail.c: Merged revisions 289104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) | 4 lines Solaris compatibility fixes Review: https://reviewboard.asterisk.org/r/942/ ........ 2010-09-28 01:10 +0000 [r289056-289058] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 289057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) | 5 lines Avoid deadlock processing incoming AOC-E messages. Deadlock avoidance for the owner channel was not done when processing incoming AOC-E messages. ........ * /, channels/chan_sip.c: Merged revisions 289054-289055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) | 1 line Break up long ast_manager_event_multichan() event lines. ........ r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 Sep 2010) | 1 line Revert stuff not ready for commit in -r289054. ........ 2010-09-27 22:03 +0000 [r289023] David Vossel * channels/chan_sip.c: For an INVITE transaction, treat all 2XX responses the same as a 200. ABE-2305 2010-09-27 19:45 +0000 [r288992-288993] Olle Johansson * channels/chan_sip.c: Formatting fixes * cdr/cdr_pgsql.c: Formating changes 2010-09-27 18:39 +0000 [r288962] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 288961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) | 5 lines Still build SIP, even if res_crypto cannot be built (use, not depend). (closes issue #18062) Reported by: a user on the mailing list ........ 2010-09-27 13:04 +0000 [r288926-288928] Russell Bryant * /, res/res_agi.c: Merged revisions 288927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288927 | russell | 2010-09-27 08:03:43 -0500 (Mon, 27 Sep 2010) | 2 lines Fix some documentation typos and spelling errors. ........ * /, res/res_agi.c: Merged revisions 288925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288925 | russell | 2010-09-27 07:42:10 -0500 (Mon, 27 Sep 2010) | 2 lines Fix a documentation spelling error. ........ 2010-09-25 07:58 +0000 [r288893] Alexandr Anikin * addons/ooh323c/src/oochannels.c: small correction for verbose print h.323 packets 2010-09-24 17:59 +0000 [r288822-288853] David Vossel * /, channels/chan_sip.c: Merged revisions 288852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) | 5 lines Append Retry-After header on 500 error response to Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........ * /, channels/chan_sip.c: Merged revisions 288821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) | 4 lines Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3. ABE-2293 ........ 2010-09-24 16:11 +0000 [r288749] Terry Wilson * channels/chan_local.c, /: Merged revisions 288748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288748 | twilson | 2010-09-24 09:02:27 -0700 (Fri, 24 Sep 2010) | 19 lines Merged revisions 288747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288747 | twilson | 2010-09-24 08:37:39 -0700 (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) | 5 lines Don't fail a masquerade if it is already being hung up This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. ........ ................ ................ 2010-09-24 13:55 +0000 [r288607-288714] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 288713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288713 | tilghman | 2010-09-24 08:54:17 -0500 (Fri, 24 Sep 2010) | 12 lines Merged revisions 288712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue #18041) Reported by: asgaroth ........ ................ * /, main/asterisk.exports.in: Merged revisions 288640 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288640 | tilghman | 2010-09-23 22:42:37 -0500 (Thu, 23 Sep 2010) | 2 lines Export timersub for platforms which do not have it ........ * /, configure, include/asterisk/autoconfig.h.in, include/asterisk/compat.h, main/strcompat.c, configure.ac, include/asterisk/channel.h, cdr/cdr_pgsql.c: Merged revisions 288638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288638 | tilghman | 2010-09-23 22:39:29 -0500 (Thu, 23 Sep 2010) | 16 lines Merged revisions 288637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500 (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines Solaris compatibility fixes ........ ................ ................ * /, CHANGES: Merged revisions 288606 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010) | 2 lines Add note about the checkhangup option of ${CHANNEL()} ........ 2010-09-23 18:08 +0000 [r288519-288573] Terry Wilson * main/manager.c, /: Merged revisions 288572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288572 | twilson | 2010-09-23 13:05:16 -0500 (Thu, 23 Sep 2010) | 2 lines Make AMI honor enabled=no ........ * channels/chan_local.c, /: Merged revisions 288507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288507 | twilson | 2010-09-22 16:18:27 -0700 (Wed, 22 Sep 2010) | 22 lines Merged revisions 288500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288500 | twilson | 2010-09-22 16:10:09 -0700 (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) | 8 lines Don't let a Local channel get bridged to itself If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 ........ ................ ................ 2010-09-22 17:50 +0000 [r288346-288419] David Vossel * /, channels/chan_sip.c: Merged revisions 288418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288418 | dvossel | 2010-09-22 12:49:56 -0500 (Wed, 22 Sep 2010) | 18 lines Merged revisions 288417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 288345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288345 | dvossel | 2010-09-22 11:59:14 -0500 (Wed, 22 Sep 2010) | 16 lines Merged revisions 288344 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ ................ ................ 2010-09-22 16:46 +0000 [r288342] Russell Bryant * /, main/asterisk.c: Merged revisions 288341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288341 | russell | 2010-09-22 11:45:18 -0500 (Wed, 22 Sep 2010) | 25 lines Merged revisions 288340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288340 | russell | 2010-09-22 11:44:13 -0500 (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf. The handling of -c and console=yes should be the same, but they were not. When you specify -c, it sets both a flag for console module and for asterisk not to fork() off into the background. The handling of console=yes only set console mode, so you would end up with a background process() trying to run the Asterisk console and freaking out since it didn't have anything to read input from. Thanks to beagles for reporting and helping debug the problem! ........ ................ ................ 2010-09-22 15:18 +0000 [r288278] Tilghman Lesher * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged revisions 288268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288268 | tilghman | 2010-09-22 10:14:02 -0500 (Wed, 22 Sep 2010) | 30 lines Merged revisions 288267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500 (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) | 9 lines Allow the encoding to be set, in case local charset does not agree with database. (closes issue #16940) Reported by: jamicque Patches: 20100827__issue16940.diff.txt uploaded by tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) | 5 lines Document addition of encoding parameter. (issue #16940) Reported by: jamicque ........ ................ ................ 2010-09-22 00:08 +0000 [r288195] Richard Mudgett * /, channels/chan_iax2.c: Merged revisions 288194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288194 | rmudgett | 2010-09-21 19:06:21 -0500 (Tue, 21 Sep 2010) | 40 lines Merged revisions 288193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel. Near the beginning of schedule_delivery(), ast_bridged_channel() is called on iaxs[fr->callno]->owner. However, the channel is not locked, which can result in ast_bridged_channel() crashing should owner->tech change to a technology that doesn't implement bridged_channel. I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since the owner lock was not held there either. Converted the existing channel deadlock avoidance to use iax2_lock_owner(). Using the new function simplified some awkward code. In the process of fixing the locking on ast_bridged_channel(), I also found a memory leak in socket_process() for v1.6.2 and v1.8. The local struct variable ies.vars is not freed on early/abnormal function exits. (closes issue #17919) Reported by: rain Patches: issue17919_v1.4.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/926/ ........ ................ ................ 2010-09-21 22:58 +0000 [r288160] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 288159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288159 | tilghman | 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) | 29 lines Merged revisions 288113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ ................ 2010-09-21 22:28 +0000 [r288158] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 288157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288157 | pabelanger | 2010-09-21 18:26:15 -0400 (Tue, 21 Sep 2010) | 15 lines Merged revisions 288147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue, 21 Sep 2010) | 9 lines Setup timer before set_config(). (closes issue #18019) Reported by: Netview Patches: issue_0018019.patch uploaded by pabelanger (license 224) Tested by: Netview ........ ................ 2010-09-21 21:04 +0000 [r288081-288083] Richard Mudgett * /, doc/tex/partymanip.tex: Merged revisions 288082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 Sep 2010) | 1 line Add note in party manipulation chapter on interception macros. ........ * apps/app_dial.c, main/channel.c, /, apps/app_queue.c: Merged revisions 288079-288080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code. ........ r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines Simplify locking code for REDIRECTING interception macro when forwarding a call. Simplified the locking code by using a local copy of the redirecting party information in app_dial.c:do_forward() and app_queue.c:wait_for_answer() for launching the REDIRECTING interception macro when a call is forwarded. Reduced the lock time of the 'o->chan' and 'in' channels. ........ 2010-09-21 20:27 +0000 [r288063] Stefan Schmidt * channels/chan_sip.c: Instead of iterate through all dialogs, add two separte container for needdestroy and rtptimeout adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks. both container will be checked on every loop of do_monitor instead of iterate through all dialogs. (closes issue #17912) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/917/ 2010-09-21 19:50 +0000 [r288008] Brett Bryant * main/channel.c, /: Merged revisions 288007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288007 | bbryant | 2010-09-21 15:48:53 -0400 (Tue, 21 Sep 2010) | 21 lines Merged revisions 288006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400 (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate memory on the first frame being queued in ast_queue_frame. (closes issue #17882) Reported by: seanbright Tested by: seanbright ........ ................ ................ 2010-09-21 19:09 +0000 [r287936] Tilghman Lesher * /, main/asterisk.c: Merged revisions 287935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287935 | tilghman | 2010-09-21 14:08:36 -0500 (Tue, 21 Sep 2010) | 16 lines Merged revisions 287934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500 (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) | 2 lines Less than zero is an error, not any non-zero value. ........ ................ ................ 2010-09-21 19:04 +0000 [r287932] Terry Wilson * main/channel.c, /: Merged revisions 287931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287931 | twilson | 2010-09-21 14:02:40 -0500 (Tue, 21 Sep 2010) | 2 lines Revert change in favor of a more targeted fix ........ 2010-09-21 18:33 +0000 [r287930] David Vossel * /, channels/chan_sip.c: Merged revisions 287929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) | 4 lines Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header. ABE-2258 ........ 2010-09-21 15:54 +0000 [r287898] Richard Mudgett * /, main/features.c: Merged revisions 287897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287897 | rmudgett | 2010-09-21 10:53:19 -0500 (Tue, 21 Sep 2010) | 1 line Cut-n-paste error in builtin_blindtransfer(). ........ 2010-09-21 15:45 +0000 [r287896] Russell Bryant * res/res_rtp_asterisk.c, main/dnsmgr.c, /, channels/chan_sip.c, main/acl.c: Merged revisions 287895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) | 10 lines Don't use ast_strdupa() from within the arguments to a function. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: russell Review: https://reviewboard.asterisk.org/r/927/ ........ 2010-09-21 15:27 +0000 [r287894] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 287893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) | 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes issue #17981) Reported by: avalentin Patches: sip-anonymous-aastra.patch uploaded by avalentin (license 1107) (plus an additional fix by me) Tested by: avalentin ........ 2010-09-21 13:45 +0000 [r287864] Russell Bryant * /, main/logger.c: Merged revisions 287863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287863 | russell | 2010-09-21 08:41:41 -0500 (Tue, 21 Sep 2010) | 2 lines Fix a regression in verbose logger processing. ........ 2010-09-21 04:39 +0000 [r287764-287834] Terry Wilson * main/channel.c, /: Merged revisions 287833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287833 | twilson | 2010-09-20 23:37:44 -0500 (Mon, 20 Sep 2010) | 3 lines Don't generate connected line buffer twice for comparison ........ * main/channel.c, /: Merged revisions 287757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287757 | twilson | 2010-09-20 18:51:38 -0500 (Mon, 20 Sep 2010) | 7 lines Avoid infinite loop with certain local channel connected line updates Compare connected line data before sending a connected line indication to avoid possible loops. Review: https://reviewboard.asterisk.org/r/932/ ........ 2010-09-21 00:04 +0000 [r287763] Brett Bryant * /, apps/app_meetme.c: Merged revisions 287760 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines Merged revisions 287759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] ........ ................ ................ 2010-09-21 00:01 +0000 [r287761-287762] Terry Wilson * /: Add alecdavis' commit to merged props * /: Add merge properties back. 2010-09-20 23:42 +0000 [r287756] Alec L Davis * main/channel.c, /: Merged revisions 287685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep 2010) | 18 lines ast_channel_masquerade: Avoid recursive masquerades. Check all 4 combinations of (original/clonechan) * (masq/masqr). Initially original->masq and clonechan->masqr were only checked. It's possible with multiple masq's planned - and not yet executed, that the 'original' chan could already have another masq'd into it - thus original->masqr would be set, that masqr would lost. Likewise for the clonechan->masq. (closes issue #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: based on bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw, alecdavis ........ 2010-09-20 23:18 +0000 [r287693] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 287683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20 Sep 2010) | 9 lines The inalarm flag was not set in sig_analog struct if the port is initially in alarm. Fixed initial inalarm value for sig_analog ports. Along with -r261007, this gets the inalarm flag in sync with chan_dahdi for sig_analog ports. (closes issue #16983) ........ 2010-09-20 22:24 +0000 [r287671] Alec L Davis * main/channel.c, /: Merged revisions 287661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep 2010) | 14 lines ast_do_masquerade. Keep channels ao2_container locked while unlink and linking channels. Previously, Masquerade would unlock 'original' and 'clonechan' and allow another masq thread to run. End result would be corrupted memory, and the frequent report 'Bad Magic Number'. (closes issue #17801,#17710) Reported by: notthematrix Patches: Based on bug17801.diff1.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/928 ........ 2010-09-20 22:16 +0000 [r287646-287648] David Vossel * main/channel.c, main/framehook.c (added), /, funcs/func_frame_trace.c (added), include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h (added): Merged revisions 287647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ ........ * /, channels/chan_sip.c: Merged revisions 287645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) | 9 lines Fixes issue with registrations not working properly with pedantic=yes. (closes issue #18017) Reported by: schmidts Patches: issues_18017_v1.diff uploaded by dvossel (license 671) Tested by: schmidts ........ 2010-09-20 21:30 +0000 [r287644] Jason Parker * /, channels/chan_skinny.c: Merged revisions 287643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287643 | qwell | 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) | 15 lines Merged revisions 287642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) | 8 lines Don't crash when parking a non-bridged call. (closes issue #17680) Reported by: jmhunter Patches: chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: jmhunter, DEA ........ ................ 2010-09-20 21:25 +0000 [r287640] Brett Bryant * /, main/logger.c: Merged revisions 287639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287639 | bbryant | 2010-09-20 17:19:12 -0400 (Mon, 20 Sep 2010) | 8 lines Fixes an error with the logger that caused verbose messages to be spammed to the screen if syslog was configured in logger.conf (closes issue #17974) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/915/ ........ 2010-09-20 15:57 +0000 [r287560] Matthew Nicholson * main/pbx.c, /: Merged revisions 287559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287559 | mnicholson | 2010-09-20 10:57:14 -0500 (Mon, 20 Sep 2010) | 21 lines Merged revisions 287558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500 (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint state changes Merged revisions 287555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines Use ast_dynamic_str when processing hint state changes (related to issue #17928) Reported by: mdu113 ........ ................ ................ 2010-09-19 16:12 +0000 [r287472] Olle Johansson * main/manager.c, /: Merged revisions 287471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287471 | oej | 2010-09-19 18:09:28 +0200 (Sön, 19 Sep 2010) | 21 lines Merged revisions 287470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön, 19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines Make sure we always free variables properly in manager originate. (closes issue #17891) reported, solved and tested by oej Review: https://reviewboard.asterisk.org/r/869/ ........ ................ ................ 2010-09-17 21:10 +0000 [r287389] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 287388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287388 | tilghman | 2010-09-17 16:08:54 -0500 (Fri, 17 Sep 2010) | 21 lines Merged revisions 287387 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines Blank columns should get set on reload, not ignored. (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) ........ ................ ................ 2010-09-17 13:38 +0000 [r287310] Matthew Nicholson * main/pbx.c, /: Merged revisions 287309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287309 | mnicholson | 2010-09-17 08:37:10 -0500 (Fri, 17 Sep 2010) | 19 lines Merged revisions 287308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500 (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed(). (related to issue #17928) Reported by: mdu113 ........ ................ ................ 2010-09-17 08:46 +0000 [r287272] Jan Kalab * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c, res/res_calendar_ews.c: Merged revisions 287269-287271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8 lines Support for HTTP redirects in calendar's URL libneon does not support HTTP redirects (3xx responses) by default. You must tell it to follow them. Also, another little unsigned int fix. (closes issue #17776) Review: https://reviewboard.asterisk.org/r/921/ ........ r287270 | pitel | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines Asterisk crashing because of double free when EWS request fails The free is done later in code. I think ast_free() should have built in checks for double free. (closes issue #17782) ........ r287271 | pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines Events are visible after they were removed from EWS calendar Because we must merge calendar even when it's empty. (closes issue #17786) ........ 2010-09-16 22:05 +0000 [r287196] Jason Parker * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287195 | qwell | 2010-09-16 17:04:38 -0500 (Thu, 16 Sep 2010) | 7 lines Don't fail when running the Debian init script directly (as one would normally do). readlink apparently returns 1 when the arg isn't a symlink, which caused the script to exit. (closes issue #17910) Reported by: wurstsalat ........ 2010-09-16 22:00 +0000 [r287194] Russell Bryant * /, configs/queues.conf.sample, apps/app_queue.c, UPGRADE-1.8.txt: Merged revisions 287193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf. Review: https://reviewboard.asterisk.org/r/922/ ........ 2010-09-16 20:08 +0000 [r287117-287121] Matthew Nicholson * main/pbx.c, /: Merged revisions 287120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287120 | mnicholson | 2010-09-16 15:07:38 -0500 (Thu, 16 Sep 2010) | 22 lines Merged revisions 287119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500 (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings. (closes issue #17928) Reported by: mdu113 Patches: 20100831__issue17928.diff.txt uploaded by tilghman (license 14) Tested by: mdu113 ........ ................ ................ * main/cdr.c, /: Merged revisions 287116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287116 | mnicholson | 2010-09-16 14:54:48 -0500 (Thu, 16 Sep 2010) | 22 lines Merged revisions 287115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500 (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines Don't stop printing cdr variables if we encounter one with a blank name or value. (closes issue #17900) Reported by: under Patches: core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ ................ ................ 2010-09-16 16:49 +0000 [r287086-287087] Olle Johansson * channels/chan_sip.c: We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incoming SIP messages. Adding error based on RFC 3398 recommendations. * main/indications.c: Add doxygen docs for indications.c 2010-09-15 22:28 +0000 [r287057] Terry Wilson * /, res/res_srtp.c: Merged revisions 287056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines Don't hang up a call on an SRTP unprotect failure Also make it more obvious when there is an issue en/decrypting. (closes issue #17563) Reported by: Alexcr Patches: res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by: twilson ........ 2010-09-15 21:00 +0000 [r287021] Jeff Peeler * /, main/features.c: Merged revisions 287020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287020 | jpeeler | 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010) | 1 line fix uninintialized variable ........ 2010-09-15 20:56 +0000 [r287018] Richard Mudgett * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ 2010-09-15 20:36 +0000 [r286939-287016] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 287015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287015 | jpeeler | 2010-09-15 15:32:52 -0500 (Wed, 15 Sep 2010) | 21 lines Merged revisions 286998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 ........ ................ ................ * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/chan_sip.c, main/features.c, CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h, configs/features.conf.sample, channels/chan_mgcp.c, include/asterisk/features.h: Merged revisions 286931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ 2010-09-15 18:30 +0000 [r286906] Richard Mudgett * channels/sig_analog.c, /: Merged revisions 286904-286905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) | 12 lines Unable to originate calls using E&M over T1. When originating a call from Unit Under Test to Reference Unit using E&M RBS signaling mode, I get the following warning message: "Ring/Off-hook in strange state 3 on channel 1". Fixed the sig_analog outgoing flag. It was never set when sig_analog was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........ r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) | 1 line Simplify some code in sig_analog. ........ 2010-09-15 13:10 +0000 [r286869] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 286868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling. This fixes a regression introduced in r274783. (closes issue #17960) Reported by: adriavidal Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mich, mnicholson, adriavidal (closes issue #17676) Reported by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ 2010-09-14 22:02 +0000 [r286835] David Vossel * /, channels/chan_sip.c: Merged revisions 286834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) | 2 lines Sets subscribed type for outgoing MWI subscriptions so correct Event header is used. ........ 2010-09-14 19:29 +0000 [r286683-286759] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 286758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286758 | mnicholson | 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) | 27 lines Merged revisions 286757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ ................ ................ * main/channel.c, /: Merged revisions 286682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286682 | mnicholson | 2010-09-14 13:04:21 -0500 (Tue, 14 Sep 2010) | 21 lines Merged revisions 286681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines Only drop duplicate answer frames if the channel is bridged. Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) ........ ................ ................ 2010-09-14 15:31 +0000 [r286648] Richard Mudgett * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Merged revisions 286647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) | 1 line Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names. ........ 2010-09-14 06:58 +0000 [r286618] Jan Kalab * /, res/res_calendar_ews.c: Merged revisions 286617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286617 | pitel | 2010-09-14 08:55:44 +0200 (Út, 14 zář 2010) | 7 lines Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar The solution is to use "global" counter of events, since we do new requests for every event and calendar sync after every request. So now we do sync only after last request. (closes issue #17877) Review: https://reviewboard.asterisk.org/r/916/ ........ 2010-09-14 05:08 +0000 [r286529-286589] Tilghman Lesher * /, contrib/realtime/mysql/voicemail_messages.sql (added), contrib/realtime/mysql/voicemail_data.sql (added): Merged revisions 286588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286588 | tilghman | 2010-09-14 00:07:16 -0500 (Tue, 14 Sep 2010) | 9 lines Merged revisions 286587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010) | 2 lines Add documentation on missing backend tables for Voicemail ........ ................ * /, main/features.c: Merged revisions 286558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286558 | tilghman | 2010-09-13 18:50:34 -0500 (Mon, 13 Sep 2010) | 9 lines Merged revisions 286557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010) | 2 lines C precedence got me ........ ................ * /, main/features.c: Merged revisions 286528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286528 | tilghman | 2010-09-13 18:12:21 -0500 (Mon, 13 Sep 2010) | 9 lines Merged revisions 286527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010) | 2 lines Refactor conversion to ast_poll() to fix callparking regression. ........ ................ 2010-09-13 22:13 +0000 [r286498] Russell Bryant * /, main/db.c: Merged revisions 286112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) | 9 lines Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was determined to be one of the most significant bottlenecks in SIP registration processing. This patch improved the speed of an astdb load test by 50000% (yes, Fifty-Thousand Percent). On this particular load test setup, this doubled the number of SIP registrations the server could handle. Review: https://reviewboard.asterisk.org/r/825/ ........ 2010-09-13 19:40 +0000 [r286458] Jason Parker * /, channels/chan_sip.c: Merged revisions 286457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286457 | qwell | 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) | 12 lines Merged revisions 286456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej ........ ................ 2010-09-13 15:53 +0000 [r286427] Richard Mudgett * configs/chan_dahdi.conf.sample, /: Merged revisions 286426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010) | 1 line Update chan_dahdi.conf.sample to reflect new libpri T309 default value. ........ 2010-09-11 17:35 +0000 [r286271-286342] Olle Johansson * main/say.c, main/app.c: Whitespace cleanup * main/features.c: Whitespace cleanup and reformatting with { and } * /, main/file.c: Merged revisions 286270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör, 11 Sep 2010) | 18 lines Merged revisions 286268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines Handle error response when we can't make file compatible Review: https://reviewboard.asterisk.org/r/911/ ........ ................ ................ * channels/chan_sip.c: Formatting changes. 2010-09-10 22:15 +0000 [r286190] Terry Wilson * channels/chan_local.c, /, funcs/func_channel.c, include/asterisk/channel.h, include/asterisk/pbx.h, include/asterisk/frame.h: Merged revisions 286189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286189 | twilson | 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) | 30 lines Merged revisions 286115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ ................ 2010-09-10 21:13 +0000 [r286121] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 286120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286120 | pabelanger | 2010-09-10 17:11:08 -0400 (Fri, 10 Sep 2010) | 18 lines Merged revisions 286117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400 (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ ................ ................ 2010-09-10 21:03 +0000 [r286119] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 286118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286118 | rmudgett | 2010-09-10 15:55:37 -0500 (Fri, 10 Sep 2010) | 25 lines Merged revisions 286116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ ................ ................ 2010-09-10 13:20 +0000 [r285993] David Ruggles * doc/externalivr.txt, CHANGES: Merged revisions 285992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep 2010) | 1 line Added missing documentation for ExternalIVR feature added in January 2010 ........ 2010-09-10 05:33 +0000 [r285932-285963] Tilghman Lesher * include/asterisk/select.h, /: Merged revisions 285962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285962 | tilghman | 2010-09-10 00:32:18 -0500 (Fri, 10 Sep 2010) | 13 lines Merged revisions 285961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines Another fix for Mac OS X. While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. ........ ................ * include/asterisk/select.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac, tests/test_poll.c: Merged revisions 285931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines Merged revisions 285930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ ................ ................ 2010-09-09 22:53 +0000 [r285820] Paul Belanger * /, codecs/gsm/Makefile: Merged revisions 285819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285819 | pabelanger | 2010-09-09 18:52:31 -0400 (Thu, 09 Sep 2010) | 22 lines Merged revisions 285818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400 (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines GCC 4.2.x optimizations result in improper behavior of GSM codec (closes issue #17688) Reported by: pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347) Tested by: mkeuter, pprindeville ........ ................ ................ 2010-09-09 20:13 +0000 [r285746] Jason Parker * main/channel.c, /: Merged revisions 285745 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285745 | qwell | 2010-09-09 15:11:06 -0500 (Thu, 09 Sep 2010) | 23 lines Merged revisions 285744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285744 | qwell | 2010-09-09 15:09:23 -0500 (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines Transmit silence when reading DTMF in ast_readstring. Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) ........ ................ ................ 2010-09-09 18:53 +0000 [r285641-285712] Brett Bryant * main/pbx.c, /: Merged revisions 285711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285711 | bbryant | 2010-09-09 14:51:52 -0400 (Thu, 09 Sep 2010) | 15 lines Merged revisions 285710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ ................ * /, res/res_musiconhold.c: Merged revisions 285640 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285640 | bbryant | 2010-09-09 13:23:28 -0400 (Thu, 09 Sep 2010) | 21 lines Merged revisions 285639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400 (Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class. (closes issue #17807) Reported by: kshumard Review: https://reviewboard.asterisk.org/r/910/ ........ ................ ................ 2010-09-08 22:15 +0000 [r285565-285569] David Vossel * /, channels/chan_sip.c: Merged revisions 285568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285568 | dvossel | 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) | 16 lines Merged revisions 285567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure. ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 285564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines Merged revisions 285563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines Fixes interoperability problems with session timer behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" header. This is not to our benefit and RFC 4028 section 7.1 even warns against it. It is possible for one endpoint to perform session-timer refreshes while the other endpoint does not support them. If in this case the end point performing the refreshing puts "timer" in the Require field during a refresh, the dialog will likely get terminated by the other end. 2. Change the behavior of 'session-timer=accept' in sip.conf (which is the default behavior of Asterisk with no session timer configuration specified) to only run session-timers as result of an incoming INVITE request if the INVITE contains an "Session-Expires" header... Asterisk is currently treating having the "timer" option in the "Supported" header as a request for session timers by the UAC. I do not agree with this. Session timers should only be negotiated in "accept" mode when the incoming INVITE supplies a "Session-Expires" header, otherwise RFC 4028 says we should treat a request containing no "Session-Expires" header as a session with no expiration. Below I have outlined some situations and what Asterisk's behavior is. The table reflects the behavior changes implemented by this patch. SITUATIONS: -Asterisk as UAS 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header 5. Outgoing INVITE: HAS "Session-Expires". Active - Asterisk will have an active refresh timer regardless if the other endpoint does. Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does. XXXXXXX - Not possible for mode. ______________________________________ |SITUATIONS | 'session-timer' MODES | |___________|________________________| | | originate | accept | |-----------|------------|-----------| |1. | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX | Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX | -------------------------------------- (closes issue #17005) Reported by: alexrecarey ........ ................ 2010-09-08 21:00 +0000 [r285534] Brett Bryant * /, apps/app_meetme.c: Merged revisions 285533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines Merged revisions 285532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq ........ ................ 2010-09-08 20:43 +0000 [r285528-285531] Jason Parker * /, include/asterisk/astobj2.h, res/res_musiconhold.c: Merged revisions 285530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285530 | qwell | 2010-09-08 15:43:10 -0500 (Wed, 08 Sep 2010) | 9 lines Merged revisions 285529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) | 1 line Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. ........ ................ * /, res/res_musiconhold.c: Merged revisions 285527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285527 | qwell | 2010-09-08 15:32:13 -0500 (Wed, 08 Sep 2010) | 15 lines Merged revisions 285526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep 2010) | 8 lines Fixes issue where moh files were no longer rescanned during a reload. (closes issue #16744) Reported by: pj Patches: 16744-reload.diff uploaded by qwell (license 4) Tested by: qwell ........ ................ 2010-09-08 07:15 +0000 [r285485] Tilghman Lesher * /, funcs/func_channel.c: Merged revisions 285484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08 Sep 2010) | 2 lines Documentation only ........ 2010-09-07 22:23 +0000 [r285394-285456] Jason Parker * /, channels/chan_sip.c: Merged revisions 285455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) | 8 lines Don't automatically add domains for wildcard bindaddrs. (closes issue #17832) Reported by: oej Patches: 17832-wildcard.diff uploaded by qwell (license 4) Tested by: qwell ........ * /, channels/chan_sip.c: Merged revisions 285369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285369 | qwell | 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) | 7 lines Add note to 'sip show settings' regarding dual-stack support, and a :: bindaddress. (closes issue #17831) Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license 4) ........ 2010-09-07 21:21 +0000 [r285374-285390] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 285386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285386 | tilghman | 2010-09-07 16:20:16 -0500 (Tue, 07 Sep 2010) | 13 lines Don't notify on attribute changes, and change how the queuing mechanism works. Fixes call spools in 1.8. (closes issue #17337) Reported by: loloski Patches: 20100827__issue17337.diff.txt uploaded by tilghman (license 14) (closes issue #17924) Reported by: mkeuter Tested by: mkeuter ........ * /, funcs/func_channel.c: Merged revisions 285373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285373 | tilghman | 2010-09-07 16:14:03 -0500 (Tue, 07 Sep 2010) | 7 lines Add CHANNEL(checkhangup) to check whether a channel is in the process of being hanged up. (closes issue #17652) Reported by: kobaz Patches: func_channel.patch uploaded by kobaz (license 834) ........ 2010-09-07 21:12 +0000 [r285372] Richard Mudgett * /, main/features.c: Merged revisions 285371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285371 | rmudgett | 2010-09-07 16:08:35 -0500 (Tue, 07 Sep 2010) | 1 line Fix cut-n-paste error. ........ 2010-09-07 20:56 +0000 [r285269-285368] Tilghman Lesher * /, pbx/pbx_config.c: Merged revisions 285367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285367 | tilghman | 2010-09-07 15:56:07 -0500 (Tue, 07 Sep 2010) | 23 lines Merged revisions 285366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500 (Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) | 9 lines Catch invalid extensions at the parser, instead of making the core deal with them. (closes issue #17794) Reported by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14) 20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14) Tested by: PavelL ........ ................ ................ * /, include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: Merged revisions 285336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285336 | tilghman | 2010-09-07 14:38:12 -0500 (Tue, 07 Sep 2010) | 2 lines Fix build on FreeBSD 8.0, take 2. ........ * /, main/poll.c: Merged revisions 285268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285268 | tilghman | 2010-09-07 14:08:09 -0500 (Tue, 07 Sep 2010) | 18 lines Merged revisions 285267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500 (Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) | 4 lines Use poll, if indicated to do so, in the ast_poll2 implementation. This fixes the unit tests on FreeBSD 8.0. ........ ................ ................ 2010-09-07 17:57 +0000 [r285199] Brett Bryant * /, apps/app_voicemail.c: Merged revisions 285197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285197 | bbryant | 2010-09-07 13:54:21 -0400 (Tue, 07 Sep 2010) | 24 lines Merged revisions 285196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400 (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] ........ ................ ................ 2010-09-07 17:55 +0000 [r285198] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 285195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285195 | rmudgett | 2010-09-07 12:47:34 -0500 (Tue, 07 Sep 2010) | 20 lines Merged revisions 285193 via svnmerge from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 285192 via svnmerge from https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........ r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does not update the caller id of the channel if a new connected number or ECT-INFORM (w/ new peer number on call transfer) is received. JIRA ABE-2502 JIRA SWP-2058 ........ ........ ................ 2010-09-06 20:10 +0000 [r285163] Russell Bryant * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 285161-285162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285161 | russell | 2010-09-06 15:10:03 -0500 (Mon, 06 Sep 2010) | 4 lines Fix libsrtp -fPIC check for when non-standard prefix is used. Thanks to loompek in #asterisk for reporting the issue and testing this patch. ........ r285162 | russell | 2010-09-06 15:10:24 -0500 (Mon, 06 Sep 2010) | 1 line regenerate configure script. ........ 2010-09-06 06:57 +0000 [r285091] Tilghman Lesher * /, BSDmakefile (added), makeopts.in: Merged revisions 285090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285090 | tilghman | 2010-09-06 01:56:07 -0500 (Mon, 06 Sep 2010) | 16 lines Merged revisions 285089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500 (Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines Silly convenience script for BSD platforms. ........ ................ ................ 2010-09-04 18:10 +0000 [r285058] Russell Bryant * /, include/asterisk/cli.h: Merged revisions 285057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285057 | russell | 2010-09-04 13:08:19 -0500 (Sat, 04 Sep 2010) | 2 lines Add a C++ compatible version of AST_CLI_DEFINE(). ........ 2010-09-03 23:23 +0000 [r285029] Terry Wilson * /, channels/chan_sip.c: Merged revisions 285017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285017 | twilson | 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) | 4 lines Call correct lock function as transferer is a sip_pvt not a channel Both functions are #defined to ao2_lock, but still... ........ 2010-09-03 22:23 +0000 [r285007] David Vossel * /, channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Merged revisions 285006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) | 9 lines Disables auth_options_request option by default. The auth_options_request option was created to do authentication on OPTIONS request just like INVITES are done. Since it has been noted that some endpoints use OPTIONS requests as a way of qualifying a peer and that a 401 authentication response could result in interoperability issues, this option has been disabled by default. ........ 2010-09-03 18:21 +0000 [r284973] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 284967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284967 | bbryant | 2010-09-03 14:19:53 -0400 (Fri, 03 Sep 2010) | 15 lines Merged revisions 284958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) | 8 lines This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response. (closes issue #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by alexkuklin (license 1115) Tested by: alexkuklin ........ ................ 2010-09-03 18:04 +0000 [r284951-284953] David Vossel * /, channels/chan_sip.c: Merged revisions 284952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) | 2 lines During OPTIONS authentication, the authpeer does not need to be returned for any reason. ........ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Merged revisions 284950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ 2010-09-03 16:42 +0000 [r284922] Terry Wilson * /, apps/app_chanspy.c: Merged revisions 284921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284921 | twilson | 2010-09-03 11:28:18 -0500 (Fri, 03 Sep 2010) | 19 lines Merged revisions 284897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they existed. ........ ................ ................ 2010-09-03 13:09 +0000 [r284851-284853] Jan Kalab * /, res/res_calendar_ews.c: Merge of strdupa() fix for calendars categories priorities * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c, include/asterisk/calendar.h, res/res_calendar_ews.c, res/res_calendar.c: Support for calendar events priorities and categories (with ISO C90 fix) See RFC 5545 ch. 3.8.1.2 and 9. (closes issue #17837) Review: https://reviewboard.asterisk.org/r/880/ 2010-09-02 21:08 +0000 [r284782] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 284779-284780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) | 8 lines Made output libpri event names if pri debugging is enabled when sig_pri processes them. * Simplified CLI "pri debug xx span xx" command code and removed redundant debugging enabled messages. * Made CLI "pri debug xx span xx" command only close the debugging log file if it was opened. ........ r284780 | rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines Simplified pri_dchannel() poll timeout duration code. ........ 2010-09-02 16:57 +0000 [r284706] David Vossel * /, channels/chan_sip.c: Merged revisions 284705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284705 | dvossel | 2010-09-02 11:56:43 -0500 (Thu, 02 Sep 2010) | 20 lines Merged revisions 284704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines Removed relatedpeer code from sip_autodestruct Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. ........ ................ ................ 2010-09-02 16:44 +0000 [r284702] Jason Parker * /, formats/format_wav.c: Merged revisions 284701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep 2010) | 8 lines Add slin16 support for format_wav (new wav16 file extension) (closes issue #15029) Reported by: andrew Patches: wav16.patch uploaded by andrew (license 240) Tested by: qwell, andrew ........ 2010-09-02 16:36 +0000 [r284700] Tilghman Lesher * /, addons/ooh323c/src/oochannels.c: Merged revisions 284696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010) | 2 lines Fixing build ........ 2010-09-02 16:35 +0000 [r284699] Richard Mudgett * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added), doc/tex/asterisk.tex: Merged revisions 284698 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) | 5 lines Added documentation for CONNECTEDLINE and REDIRECTING functions. (closes issue #17808) Reported by: jtodd Review: https://reviewboard.asterisk.org/r/875/ ........ 2010-09-02 16:12 +0000 [r284598-284667] Tilghman Lesher * channels/chan_usbradio.c, /: Merged revisions 284666 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284666 | tilghman | 2010-09-02 11:11:15 -0500 (Thu, 02 Sep 2010) | 9 lines Merged revisions 284665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) | 2 lines Fixing build. ........ ................ * /, apps/app_queue.c: Merged revisions 284632 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines Merged revisions 284631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines Don't reset queue stats on a module reload. (closes issue #17535) Reported by: raarts Patches: 20100819__issue17535.diff.txt uploaded by tilghman (license 14) ........ ................ * /, channels/chan_sip.c, channels/chan_agent.c, channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c, apps/app_followme.c, apps/app_speech_utils.c, main/loader.c, pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c, include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_stack.c, channels/chan_mgcp.c, apps/app_adsiprog.c, apps/app_voicemail.c: Merged revisions 284610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ * main/stun.c, res/res_ais.c, /, include/asterisk/autoconfig.h.in, configure.ac, channels/console_video.c, include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c, main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h (added), channels/chan_phone.c, channels/chan_misdn.c, configure, main/features.c, include/asterisk/poll-compat.h, tests/test_poll.c (added), addons/ooh323c/src/oochannels.c, main/asterisk.c, addons/ooh323c/src/ooSocket.h: Merged revisions 284597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines Merged revisions 284593,284595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after last commit ................ ................ 2010-09-01 21:48 +0000 [r284562] David Vossel * /, channels/chan_sip.c: Merged revisions 284561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines During request to dialog matching, verify init_ruri is present before comparing. During request to dialog matching, we attempt a best effort routine for fork detection which requires several elements to be in place. The dialog's initial request uri is one of those elements. Since it is best effort, if the init_ruri is not present for some reason we can not proceed with that routine. ........ 2010-09-01 18:52 +0000 [r284479] Terry Wilson * res/res_rtp_asterisk.c, include/asterisk/res_srtp.h, main/rtp_engine.c, /, channels/chan_sip.c, res/res_srtp.c: Merged revisions 284477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ 2010-09-01 18:19 +0000 [r284440-284474] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 284473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284473 | tilghman | 2010-09-01 13:16:37 -0500 (Wed, 01 Sep 2010) | 12 lines Merged revisions 284472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) | 5 lines Don't warn on floats and timestamps (closes issue #17082) Reported by: coolmig ........ ................ * /, channels/chan_sip.c: Merged revisions 284415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284415 | tilghman | 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) | 21 lines Merged revisions 284399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ ................ ................ 2010-08-31 19:01 +0000 [r284315-284319] Leif Madsen * /, configs/say.conf.sample: Merged revisions 284318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284318 | lmadsen | 2010-08-31 14:00:15 -0500 (Tue, 31 Aug 2010) | 22 lines Merged revisions 284317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500 (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines Update say.conf.sample to match the rules in say.c (closes issue #17835) Reported by: RoadKill Patches: say.conf.sample.patch.rules uploaded by RoadKill (license 933) Tested by: RoadKill ........ ................ ................ * channels/chan_sip.c: Add trustrpid and sendrpid global values to 'sip show settings' (closes issue #17860) Reported by: jtodd Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10) Tested by: lmadsen, russell 2010-08-30 22:30 +0000 [r284282] Tilghman Lesher * /, apps/app_festival.c: Merged revisions 284281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284281 | tilghman | 2010-08-30 17:28:47 -0500 (Mon, 30 Aug 2010) | 18 lines Merged revisions 284280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines Fix 3 coding errors: 1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez ........ ................ 2010-08-30 09:32 +0000 [r284189-284248] Olle Johansson * main/file.c: Small doxygen fix and doc addition * main/say.c: Clean upp doxygen documentation * include/asterisk/say.h: Doxygen formatting You can't write "same as above" in hypertext documentation. Above doesn't make sense in hyperspace. * apps/app_playback.c: Add doxygen documentation 2010-08-29 07:06 +0000 [r284097-284159] Tilghman Lesher * /, configs/res_curl.conf.sample (added): Merged revisions 284158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284158 | tilghman | 2010-08-29 02:05:27 -0500 (Sun, 29 Aug 2010) | 2 lines Missed adding this file ........ * /, sounds: Merged revisions 284127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284127 | tilghman | 2010-08-29 00:17:37 -0500 (Sun, 29 Aug 2010) | 2 lines Also ignore the checksums ........ * cel/cel_odbc.c (added), /, configs/cel_adaptive_odbc.conf.sample (removed), configs/cel_odbc.conf.sample (added), cel/cel_adaptive_odbc.c (removed): Merged revisions 284096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284096 | tilghman | 2010-08-28 21:51:14 -0500 (Sat, 28 Aug 2010) | 3 lines Rename CEL adaptive driver to plain driver, since there isn't another ODBC driver (and the other CEL drivers have adaptive capabilities, anyway). ........ 2010-08-28 21:30 +0000 [r284066] Russell Bryant * main/manager.c, /: Merged revisions 284065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines Be more flexible with whitespace on AMI action headers. Previously, this code required exactly one space to be after the ':' in headers for an AMI action. This now makes whitespace optional, and allows whitespace that is there to vary in amount. (closes issue #17862) Reported by: cmoye Patches: manager.c.patch_trunk uploaded by cmoye (license 858) manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by: cmoye ........ 2010-08-27 22:39 +0000 [r284033] David Vossel * /, channels/chan_sip.c: Merged revisions 284032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284032 | dvossel | 2010-08-27 17:37:11 -0500 (Fri, 27 Aug 2010) | 21 lines Merged revisions 284002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ ................ ................ 2010-08-27 21:50 +0000 [r283958] Russell Bryant * /, pbx/pbx_realtime.c: Merged revisions 283951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283951 | russell | 2010-08-27 16:33:55 -0500 (Fri, 27 Aug 2010) | 2 lines Print exten@context:priority in verbose messages from pbx_realtime. ........ 2010-08-27 20:32 +0000 [r283883] Jason Parker * res/res_config_odbc.c, /, main/config.c, addons/res_config_mysql.c: Merged revisions 283882 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283882 | qwell | 2010-08-27 15:31:55 -0500 (Fri, 27 Aug 2010) | 22 lines Merged revisions 283881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283881 | qwell | 2010-08-27 15:30:27 -0500 (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines Fix issue with decoding ^-escaped characters in realtime. (closes issue #17790) Reported by: denzs Patches: 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell, denzs ........ ................ ................ 2010-08-27 14:01 +0000 [r283803] Olle Johansson * main/manager.c: Doxygen formatting changes 2010-08-26 23:51 +0000 [r283771] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 283770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26 Aug 2010) | 8 lines Convert MOH to use generic timers. (closes issue #17726) Reported by: lmadsen Patches: 20100825__issue17726__2.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-08-26 15:28 +0000 [r283693] David Vossel * /, channels/chan_sip.c: Merged revisions 283692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283692 | dvossel | 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) | 32 lines Merged revisions 283691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ ................ ................ 2010-08-26 13:28 +0000 [r283628-283660] Russell Bryant * /, res/res_odbc.c: Merged revisions 283659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010) | 2 lines Slight improvement to a debug message. ........ * Makefile, /, keys/iaxtel.pub (removed), keys/freeworlddialup.pub (removed): Merged revisions 283629 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283629 | russell | 2010-08-26 07:48:45 -0500 (Thu, 26 Aug 2010) | 2 lines Remove public keys that are no longer useful. ........ * /, configs/manager.conf.sample: Merged revisions 283627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283627 | russell | 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010) | 2 lines Move httptimeout out from in between port and bindaddr. ........ 2010-08-25 22:59 +0000 [r283596] David Vossel * /, channels/chan_sip.c: Merged revisions 283595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines Merged revisions 283594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines Add to and from tags to NOTIFY dialog-info xml body so pickup can occur. When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. ........ ................ 2010-08-25 16:14 +0000 [r283562] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 283561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) | 5 lines Initialize connect timeout on each time through the loop. (closes issue #17911) Reported by: wurstsalat ........ 2010-08-25 15:56 +0000 [r283560] David Vossel * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 283559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines Merged revisions 283558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ ................ 2010-08-25 14:55 +0000 [r283528] Russell Bryant * /, channels/chan_sip.c: Merged revisions 283527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) ........ 2010-08-24 20:42 +0000 [r283495] Damien Wedhorn * channels/chan_skinny.c: Ignore redial hard button when no previous number. (closes issue #17887) Reported by: salecha Patches: skinny.redial.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha 2010-08-24 20:36 +0000 [r283494] David Vossel * /, configs/sip.conf.sample, UPGRADE-1.8.txt, channels/sip/include/sip.h: Merged revisions 283493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ 2010-08-24 18:58 +0000 [r283458] Leif Madsen * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions 283457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines Fix issue where TOS is no longer set on RTP packets. Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk. (closes issue #17890) Reported by: elguero Patches: qos_18.diff uploaded by elguero (license 37) Review: https://reviewboard.asterisk.org/r/868 ........ 2010-08-24 18:45 +0000 [r283383-283456] David Vossel * res/res_stun_monitor.c: This fix downgrades the ERROR message indicating no res_stun_monitor.conf to a WARNING message. * /, channels/chan_sip.c: Merged revisions 283382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines Merged revisions 283381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ ................ ................ 2010-08-24 12:51 +0000 [r283351] Russell Bryant * /, funcs/func_odbc.c: Merged revisions 283350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) | 2 lines Don't attempt to release a NULL ODBC handle. ........ 2010-08-23 21:35 +0000 [r283320] Tilghman Lesher * /, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, cel/cel_adaptive_odbc.c: Merged revisions 283319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283319 | tilghman | 2010-08-23 16:33:47 -0500 (Mon, 23 Aug 2010) | 9 lines Merged revisions 283318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010) | 2 lines CDR drivers depend upon res_odbc, not directly on the ODBC libraries ........ ................ 2010-08-23 20:50 +0000 [r283287-283289] Damien Wedhorn * channels/chan_skinny.c: Hack to allow easy debugging of skinny in trunk. * channels/chan_skinny.c: Add additional AST_CONTROL_ states to control2str. * channels/chan_skinny.c: Fixes display issues on 7910 and older phones. Also correct the callinfo provided in skinny_answer. (closes issue #17876) Reported by: salecha Patches: skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA 2010-08-23 13:35 +0000 [r283178-283242] Russell Bryant * configs/cel.conf.sample, /: Merged revisions 283241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23 Aug 2010) | 2 lines Add sample configuration for cel_radius. ........ * /, include/asterisk/cel.h, main/cel.c: Merged revisions 283230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values. Really, having 2 enums for this is silly and error prone, demonstrated by the crash that I hit because there was an assumption in the code that the values in each matched up. However, this is a quick fix to get them to match up so it will work. ........ * /, main/cel.c: Merged revisions 283209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) | 2 lines Don't blow up on an invalid AMA flag. ........ * /, configs/cel_custom.conf.sample: Merged revisions 283207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010) | 2 lines Tack on ${eventextra} to the sample cel_custom.conf. ........ * /, configs/cel_custom.conf.sample: Merged revisions 283177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010) | 2 lines Cut down on excessive quotation. ........ 2010-08-23 12:09 +0000 [r283176] Tilghman Lesher * /, res/res_stun_monitor.c: Merged revisions 283175 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 Aug 2010) | 2 lines Don't fail to start if the config file is missing. ........ 2010-08-23 11:59 +0000 [r283174] Russell Bryant * /, configs/cel_custom.conf.sample: Merged revisions 283173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) | 5 lines Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list the special CEL variables that are available for use in the mapping. ........ 2010-08-20 15:39 +0000 [r283051] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 283050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines Merged revisions 283049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ ................ ................ 2010-08-20 12:45 +0000 [r282980-283015] Russell Bryant * /, configs/cel_adaptive_odbc.conf.sample: Merged revisions 283013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010) | 2 lines Fix a typo in a column name. ........ * /, apps/app_celgenuserevent.c: Merged revisions 282979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010) | 2 lines Add an argument missing from the CELGenUserEvent documentation. ........ * channels/chan_multicast_rtp.c, /: Merged revisions 282638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) | 4 lines Split _all_ arguments before parsing them. This fixes multicast RTP paging using linksys mode. ........ 2010-08-19 21:08 +0000 [r282892-282896] David Vossel * /, channels/chan_sip.c: Merged revisions 282895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 282891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282891 | dvossel | 2010-08-19 15:34:41 -0500 (Thu, 19 Aug 2010) | 11 lines Merged revisions 282890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines fixes sip peer memory leaks in the peer_by_ip table (issue #17798) ........ ................ 2010-08-19 20:02 +0000 [r282861] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 282860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282860 | mnicholson | 2010-08-19 15:01:11 -0500 (Thu, 19 Aug 2010) | 30 lines Merged revisions 282859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ ................ ................ 2010-08-19 14:46 +0000 [r282827] Tilghman Lesher * main/netsock2.c, /: Merged revisions 282826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) | 2 lines Only output debugging if the debug level is on. ........ 2010-08-19 12:13 +0000 [r282798] Russell Bryant * include/asterisk/cel.h: Add a todo item for CEL. 2010-08-19 02:20 +0000 [r282751] Terry Wilson * /, configs/sip.conf.sample: Merged revisions 282740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines Merged revisions 282730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines Add some documentation about codec negotiation to sip.conf ........ ................ ................ 2010-08-18 21:34 +0000 [r282701] Damien Wedhorn * channels/chan_skinny.c: Cleanup: consolidate offhook (new call). Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one. (closes issue #17812) Reported by: wedhorn Patches: cleanup.stateoffhook.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA 2010-08-18 15:35 +0000 [r282673] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /: Merged revisions 282671-282672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) | 1 line Use the correct operator when calculating the PRI span devstate. ........ r282672 | rmudgett | 2010-08-18 10:28:27 -0500 (Wed, 18 Aug 2010) | 1 line Use the correct type for aoce_delayhangup bit field. ........ 2010-08-18 13:11 +0000 [r282640] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 282639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests. This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received. (issue #17486) Reported by: davidw Tested by: mnicholson (issue #12713) Reported by: davidw Review: https://reviewboard.asterisk.org/r/860/ ........ 2010-08-18 07:50 +0000 [r282609] Tilghman Lesher * /, channels/sig_pri.c: Merged revisions 282608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282608 | tilghman | 2010-08-18 02:49:04 -0500 (Wed, 18 Aug 2010) | 16 lines Merged revisions 282607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes. (closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ ................ 2010-08-17 21:37 +0000 [r282544-282578] David Vossel * /, channels/chan_sip.c: Merged revisions 282577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282577 | dvossel | 2010-08-17 16:36:57 -0500 (Tue, 17 Aug 2010) | 16 lines Merged revisions 282576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines fixes no default transport for temp peer creation in chan_sip (closes issue #17829) Reported by: falves11 Patches: issue_17829.rev1.txt uploaded by russell (license 2) issue_17829.diff uploaded by dvossel (license 671) Tested by: falves11 ........ ................ * /, channels/chan_iax2.c: Merged revisions 282545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 Aug 2010) | 6 lines ACCEPT message should respond with the new FORMAT2 ie (closes issue #17804) Reported by: tpanton ........ * /, include/asterisk/unaligned.h: Merged revisions 282543 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) | 4 lines fixes truncated uint64_t value in put_unaligned_uint64_t() function (issue #17804) ........ 2010-08-16 20:40 +0000 [r282502] Terry Wilson * main/channel.c, /: Merged revisions 282468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines Merged revisions 282467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines Send a SRCCHANGE indication when we masquerade Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ ........ ................ ................ 2010-08-16 18:02 +0000 [r282471] Leif Madsen * doc/tex/sounds.tex (added), /, doc/tex/asterisk.tex: Merged revisions 282470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines Merged revisions 282469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines Add information about creating sounds files using the sounds tools publically available so that others can create their own sounds prompts using the same tools we use to generate sounds releases. This allows people creating their own prompts to sound consistent with the prompts available from the open source project. SWP-595 ........ ................ 2010-08-15 13:08 +0000 [r282397] Tzafrir Cohen * utils/muted.c, configure, main/Makefile, configure.ac, main/acl.c, channels/chan_oss.c, main/netsock.c: Support for GNU/kFreeBSD kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING This patch gets Asterisk close to building on Debian kFreeBSD i386, mainly by adding an extra test for __GLIBC__ in one or two (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support (and support for chan_vpb) was not tested. Review: https://reviewboard.asterisk.org/r/858/ 2010-08-14 04:58 +0000 [r282367] Tilghman Lesher * /, include/asterisk/sched.h, channels/chan_iax2.c: Merged revisions 282366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines Fix our FRACKing issue with chan_iax2 a different way. Review: https://reviewboard.asterisk.org/r/861/ ........ 2010-08-13 23:57 +0000 [r282335] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 282334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines PRI CCSS may use a stale dial string for the recall dial string. If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. ........ 2010-08-13 22:27 +0000 [r282237-282304] David Vossel * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, UPGRADE-1.8.txt: Merged revisions 282302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ * /, CHANGES: Merged revisions 282271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines res_stun_monitor and corresponding options CHANGES documentation ........ * configs/iax.conf.sample, /, channels/chan_sip.c, include/asterisk/event_defs.h, res/res_stun_monitor.c (added), configs/res_stun_monitor.conf.sample (added), configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions 282269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines res_stun_monitor for monitoring network changes behind a NAT device Review: https://reviewboard.asterisk.org/r/854 ........ * /, channels/chan_sip.c: Merged revisions 282236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines Merged revisions 282235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines only do magic pickup when notifycid is enabled A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set... but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (closes issue #17633) Reported by: urosh Patches: chan_sip.txt uploaded by urosh (license ) blf_cid_issue.diff uploaded by dvossel (license 671) Tested by: dvossel, urosh, okrief, alecdavis ........ ................ 2010-08-13 16:08 +0000 [r282202] Terry Wilson * /, configure, configure.ac: Merged revisions 282200-282201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines Detect when libsrtp cannot be linked in a shared library The libsrtp build system currently does not produce a shared library or a static library compiled with -fPIC, so on 64-bit systems it is possible that we will get a compile error if libsrtp is installed and res_srtp is selected in menuselect. This patch attempts to detect this situation and provide the user with instructions to work around the problem. ........ r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines Whitespace fix :-/ ........ 2010-08-12 22:52 +0000 [r282132] Jason Parker * /, pbx/pbx_config.c: Merged revisions 282131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282131 | qwell | 2010-08-12 17:51:44 -0500 (Thu, 12 Aug 2010) | 16 lines Merged revisions 282130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282130 | qwell | 2010-08-12 17:50:54 -0500 (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line Register CLI commands before parsing config, in case there is a config error. ........ ................ ................ 2010-08-12 22:10 +0000 [r282099] Richard Mudgett * /, main/ccss.c, include/asterisk/ccss.h: Merged revisions 282098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines Separate call completion config parameter allocation and default initialization. If you ever have a need to reset the call completion config parameters to defaults, now you can. And no Virginia, C++ idioms do not always work in C. ........ 2010-08-12 20:44 +0000 [r282067] Russell Bryant * /, CHANGES, main/cli.c: Merged revisions 282066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines Add a "core reload" CLI command. Review: https://reviewboard.asterisk.org/r/859/ ........ 2010-08-12 20:17 +0000 [r282048] David Vossel * /, main/translate.c, CHANGES, include/asterisk/translate.h, main/cli.c: Merged revisions 282047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines improved translation paths for wideband codecs The problem I'm addressing is that Asterisk's current method of building the least cost translation paths between codecs does not take into account sample rate. For instance, it was possible for siren14 (a 32khz codec), to contain the a translation path to siren7 (a 16khz audio codec) that goes through slin at 8khz. In this case Asterisk takes a 32khz codec, down samples it to 8khz and then up samples it to 16khz which is terrible regardless if it is computationally less expensive. This patch now builds translation paths that give priority to maintaining the best possible sample rate before taking into consideration computational cost. This patch also adds cli commands to expose what translation paths are actually being used. Changes: 1. Translation paths will never contain a step that changes the sample rate unless absolutely necessary. 2. When choosing the best codec to make two channels compatible. Shared codecs with the highest sample rate are given priority. 3. A new cli command to show all translation paths available for a specific codec 'core show translation paths [codec name]' has been added. 4. 'core show translation' which displays the translation matrix now includes the new higher bit audio codecs in the table. 5. 'core show channel [channel name]' now displays the translation paths if translation is used. (closes issue #16841) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/842/ ........ 2010-08-12 18:04 +0000 [r281983-282016] Russell Bryant * main/pbx.c, /: Merged revisions 282015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) | 2 lines Put back pointer value output for ast_debug(), such that it is only removed for verbose output. ........ * main/pbx.c, /: Merged revisions 281982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines Remove debugging output from verbose messages. Pointer values to internal objects is not terribly useful to users in the verbose messages about adding extensions and contexts. ........ 2010-08-12 03:08 +0000 [r281914] Jeff Peeler * main/channel.c, /: Merged revisions 281913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines Merged revisions 281912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines Ensure SSRC is changed when media source is changed to resolve audio delay. This change causes the SSRC to change right before the channels are bridged, which is what used to happen. It seems that fixes were made to attempt limiting SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC with this change. There are two other control frames sent in ast_channel_bridge that probably should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change up to the discretion of resolving issue #17007. For reference - old review implementing new control frame SRCCHANGE: https://reviewboard.asterisk.org/r/540 (closes issue #17404) Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler (license 325) Tested by: sdolloff ........ ................ ................ 2010-08-11 21:13 +0000 [r281877] Leif Madsen * /, configs/say.conf.sample: Merged revisions 281875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281875 | lmadsen | 2010-08-11 16:12:13 -0500 (Wed, 11 Aug 2010) | 21 lines Merged revisions 281873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500 (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes issue #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk uploaded by RoadKill (license 933) ........ ................ ................ 2010-08-11 21:12 +0000 [r281876] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 281874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug 2010) | 10 lines handle all possible responses to REFER requests (closes issue #17486) Reported by: davidw Patches: Issue17486-counterbid.diff.txt uploaded by davidw (license 780) Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/ ........ 2010-08-11 20:38 +0000 [r281871] Richard Mudgett * channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions 281870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) | 4 lines Fix a call to analog_set_pulsedial() not setting 0 or 1 only. * Also a couple minor tweaks. ........ 2010-08-11 17:55 +0000 [r281765] Leif Madsen * /, configs/say.conf.sample: Merged revisions 281764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281764 | lmadsen | 2010-08-11 12:54:56 -0500 (Wed, 11 Aug 2010) | 21 lines Merged revisions 281763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500 (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines Allow say.conf to handle large numbers ending with multiple zeros. (closes issue #17833) Reported by: RoadKill Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933) ........ ................ ................ 2010-08-11 17:29 +0000 [r281761] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 281760 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug 2010) | 4 lines Avoid a deadlock in add_header_max_forwards(). Related to r276951 ........ 2010-08-11 15:20 +0000 [r281726] Tilghman Lesher * /, apps/app_readexten.c: Merged revisions 281723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281723 | tilghman | 2010-08-11 10:18:40 -0500 (Wed, 11 Aug 2010) | 14 lines Merged revisions 281722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) | 7 lines Only set status TIMEOUT, if we have no digits. (closes issue #15188) Reported by: jcovert Patches: app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551) ........ ................ 2010-08-11 13:31 +0000 [r281688] * main/netsock2.c, /, include/asterisk/netsock2.h, configs/sip.conf.sample, channels/sip/config_parser.c: Merged revisions 281687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) | 9 lines Fix parsing of IPv6 address literals in outboundproxy (closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej ........ 2010-08-10 21:50 +0000 [r281530-281651] Russell Bryant * /, configs/sip.conf.sample, UPGRADE-1.8.txt, channels/sip/include/sip.h: Merged revisions 281650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines Change the default value for alwaysauthreject in sip.conf to "yes". (closes issue #17756) Reported by: oej ........ * /, main/sched.c: Merged revisions 281575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281575 | russell | 2010-08-10 13:05:07 -0500 (Tue, 10 Aug 2010) | 16 lines Merged revisions 281574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) | 9 lines Don't move the time threshold for running scheduled events on every iteration. Instead, only calculate the time threshold each time ast_sched_runq() is called. (closes issue #17742) Reported by: schmidts Patches: sched.c.patch uploaded by schmidts (license 1077) ........ ................ * apps/app_dial.c, /: Merged revisions 281568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281568 | russell | 2010-08-10 12:48:42 -0500 (Tue, 10 Aug 2010) | 22 lines Merged revisions 281567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281567 | russell | 2010-08-10 12:47:13 -0500 (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 281532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010) | 8 lines Ensure that the proper external address is used for the RTP destination. (closes issue #17044) Reported by: ebroad Tested by: ebroad Review: https://reviewboard.asterisk.org/r/566/ ........ * /, main/cli.c: Merged revisions 281529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) | 8 lines Resolve a problem with channel name tab completion. Hitting tab without typing any part of a channel name resulted in no results. This now results in getting a full list of active channels, just as it did in previous versions of Asterisk. Review: https://reviewboard.asterisk.org/r/818/ ........ 2010-08-10 07:26 +0000 [r281498] TransNexus OSP Development * apps/app_osplookup.c: Fixed the issue caused by EXTEN including user parameters. 2010-08-09 23:04 +0000 [r281467] Jeff Peeler * channels/chan_local.c, /: Merged revisions 281466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281466 | jpeeler | 2010-08-09 18:04:02 -0500 (Mon, 09 Aug 2010) | 2 lines Add some more stuff to copy from 281429. ........ 2010-08-09 20:49 +0000 [r281433] David Vossel * /, channels/chan_sip.c: Merged revisions 281432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281432 | dvossel | 2010-08-09 15:47:53 -0500 (Mon, 09 Aug 2010) | 20 lines Merged revisions 281430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines fixes SIP peers memory leak We zeroed out the peer's addr before it was removed from the peers_by_ip container. This made it impossible to be removed from the container as the addr is the key used by the container to find the peer. (closes issue #17774) Reported by: kkm Patches: 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) ........ ................ 2010-08-09 20:46 +0000 [r281431] Jeff Peeler * channels/chan_local.c, /: Merged revisions 281429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281429 | jpeeler | 2010-08-09 15:43:54 -0500 (Mon, 09 Aug 2010) | 27 lines Merged revisions 281391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines Prevent loss of Caller ID information set on local channel after masquerade. Caller ID set on the channel before a masquerade occurs when using a local channel would cause the information to be lost. The problem was that the information was set on a channel destined to be hung up. The somewhat confusing fix is to detect if any Caller ID has been set on the channel and if so preswap the Caller ID data so that basically the masquerade puts the data back. (closes issue #17138) Reported by: kobaz Review: https://reviewboard.asterisk.org/r/847/ ........ ................ ................ 2010-08-09 14:52 +0000 [r281359] Matthew Nicholson * /, res/res_fax.c: Merged revisions 281358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug 2010) | 4 lines Validate minrate, maxrate, and modem settings before attempting a fax session. FAX-224 ........ 2010-08-09 14:32 +0000 [r281357] * /, configs/sip.conf.sample: Merged revisions 281356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 (Mon, 09 Aug 2010) | 2 lines Added comment about IPv4-mapped IPv6 addresses and the output of netstat. ........ 2010-08-09 12:52 +0000 [r281295-281326] Russell Bryant * /, configs/cdr.conf.sample: Merged revisions 281325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281325 | russell | 2010-08-09 07:51:43 -0500 (Mon, 09 Aug 2010) | 2 lines Add a couple of default values to the documentation of cdr.conf. ........ * /, configs/cdr.conf.sample: Merged revisions 281294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09 Aug 2010) | 5 lines Reorder some options in cdr.conf.sample. Put all of the options that affect the contents of CDRs together, instead of having the batch mode options in the middle of them. ........ 2010-08-07 22:36 +0000 [r281226-281257] Damien Wedhorn * channels/chan_skinny.c: Fix up handling and indications during transfer. Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up. (closes issue #17692) Reported by: jmhunter Patches: chan_skinny-transfer-v4.txt uploaded by DEA (license 3) skinnytransfver.v8.diff uploaded by wedhorn (license 30) Tested by: jmhunter, salecha, wedhorn Review: NA * channels/chan_skinny.c: Move call answering stuff into new setsubstate_connected. Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected. (closes issue #17772) Reported by: wedhorn Patches: cleanup.stateconnected2.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha Review: NA * channels/chan_skinny.c: Start rtp on answer before the answer is queued (closes issue #17770) Reported by: salecha Patches: skinny.answercrash.diff uploaded by wedhorn (license 30) Tested by: salecha Review: NA 2010-08-06 18:58 +0000 [r281086] Tilghman Lesher * /, main/utils.c: Merged revisions 281085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010) | 8 lines Fix alignment of stringfields on the SPARC architecture (closes issue #17789) Reported by: Ian Mason Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman (license 14) Tested by: Ian_Mason ........ 2010-08-05 13:19 +0000 [r281054] Russell Bryant * main/cdr.c, /: Merged revisions 281052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281052 | russell | 2010-08-05 08:16:11 -0500 (Thu, 05 Aug 2010) | 16 lines Merged revisions 281051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) | 9 lines Cleanup default option value handling for cdr.conf [general]. The default values would differ depending on whether or not cdr.conf exists. That is no longer the case. Apply a default value to the unanswered option. Define all default values as named constants. ........ ................ 2010-08-05 07:47 +0000 [r280985] Tilghman Lesher * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 280984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280984 | tilghman | 2010-08-05 02:46:36 -0500 (Thu, 05 Aug 2010) | 22 lines Merged revisions 280983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500 (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines Change context lock back to a mutex, because functionality depends upon the lock being recursive. (closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ ................ 2010-08-04 15:22 +0000 [r280910] Matthew Nicholson * /, res/res_fax.c: Merged revisions 280909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug 2010) | 2 lines Initialize FAXOPT() status variables in sendfax and receivefax instead of when the details structure is created. ........ 2010-08-04 14:05 +0000 [r280810-280880] Tilghman Lesher * /, channels/chan_mgcp.c: Merged revisions 280879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 Aug 2010) | 14 lines Check cur value before attempting a deref. (closes issue #17775) Reported by: svinson Patches: 20100804__issue17775.diff.txt uploaded by tilghman (license 14) Tested by: svinson (closes issue #17743) Reported by: tgruenberg Patches: 20100804__issue17775.diff.txt uploaded by tilghman (license 14) Tested by: tgruenberg ........ * /, funcs/func_strings.c, CHANGES: Merged revisions 280809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) | 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth Patches: svn-279754.diff uploaded by gareth (license 208) Tested by: gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ ........ 2010-08-03 19:59 +0000 [r280745-280780] * /, channels/chan_sip.c: Merged revisions 280778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 Aug 2010) | 9 lines Fixed IPv6-related SIP parsing bugs. (closes issue #17663) Reported by: oej Patches: diff uploaded by sperreault (license 252) diff2 uploaded by sperreault (license 252) get_domain.diff uploaded by sperreault (license 252) ........ * /, configs/sip.conf.sample: Merged revisions 280777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 (Tue, 03 Aug 2010) | 8 lines Better documentation related to IPv6. (closes issue #17737) Reported by: oej Patches: doc.diff uploaded by sperreault (license 252) Tested by: mmichelson ........ * contrib/realtime/mysql/voicemail.sql, channels/chan_sip.c, contrib/realtime/mysql/iaxfriends.sql, contrib/realtime/mysql/meetme.sql, contrib/realtime/postgresql/realtime.sql, configs/sip.conf.sample, contrib/realtime/mysql/sipfriends.sql: Reverted r280706 and r280707. Will commit in branch 1.8 and merge to trunk properly. 2010-08-03 18:50 +0000 [r280743] Russell Bryant * contrib/scripts/get_mp3_source.sh (added), /, addons/Makefile, addons/mp3 (removed): Merged revisions 280742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010) | 4 lines Remove the MP3 decoder source code and replace it with a small shell script. Review: https://reviewboard.asterisk.org/r/836/ ........ 2010-08-03 18:43 +0000 [r280741] Tilghman Lesher * /, doc/asterisk.8, doc/Makefile (added), doc/asterisk.sgml: Merged revisions 280740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280740 | tilghman | 2010-08-03 13:42:24 -0500 (Tue, 03 Aug 2010) | 9 lines Merged revisions 280739 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines Document -B and -W flags and regenerate manpage from sgml ........ ................ 2010-08-03 16:52 +0000 [r280706-280707] * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes issue #17663) Reported by: oej Patches: diff uploaded by sperreault (license 252) diff2 uploaded by sperreault (license 252) get_domain.diff uploaded by sperreault (license 252) * configs/sip.conf.sample: Better documentation related to IPv6. 2010-08-02 21:28 +0000 [r280629-280673] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 280672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280672 | tilghman | 2010-08-02 16:27:25 -0500 (Mon, 02 Aug 2010) | 9 lines Merged revisions 280671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) | 2 lines Allow the pipe, but also allow the comma ........ ................ * /, main/Makefile: Merged revisions 280628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280628 | tilghman | 2010-08-02 09:41:46 -0500 (Mon, 02 Aug 2010) | 2 lines Make this a little more deterministic... we want the latest value, not just a 1 somewhere. ........ 2010-08-02 14:30 +0000 [r280627] David Vossel * channels/chan_sip.c: if totag is not present for an ACK request, do not send an error response 2010-08-02 14:28 +0000 [r280626] Tilghman Lesher * /, main/Makefile: Merged revisions 280624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280624 | tilghman | 2010-08-02 09:27:20 -0500 (Mon, 02 Aug 2010) | 2 lines Apparently, the values in makeopts are sometimes 1:1 and sometimes 1. Compensate for this. ........ 2010-07-30 09:12 +0000 [r280589] Damien Wedhorn * channels/chan_skinny.c: Cleanup transmit_ for handle_register and keepalives Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session). (closes issue #16980) Reported by: wedhorn Patches: skinny-clean06b.diff uploaded by wedhorn (license 30) Tested by: wedhorn, DEA Review: NA 2010-07-29 21:08 +0000 [r280559] Matthew Nicholson * /, res/res_fax.c: Merged revisions 280557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280557 | mnicholson | 2010-07-29 16:07:21 -0500 (Thu, 29 Jul 2010) | 4 lines Fix regression introduced in r1664. Give the fax stack time to shutdown and populate the FAXOPT output variables. FAX-222 ........ 2010-07-29 21:06 +0000 [r280555] Paul Belanger * CHANGES, channels/chan_iax2.c: PeerStatus now includes Address and Port (closes issue #17730) Reported by: jkroon Patches: iax2-peerstate-address.patch uploaded by jkroon (license 714) Tested by: lmadsen 2010-07-29 20:44 +0000 [r280553] David Vossel * /, channels/chan_sip.c: Merged revisions 280552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280552 | dvossel | 2010-07-29 15:43:47 -0500 (Thu, 29 Jul 2010) | 17 lines Merged revisions 280551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) | 11 lines fixes wrong SRV query for TLS connection (closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger ........ ................ 2010-07-29 20:36 +0000 [r280550] Russell Bryant * /, configs/ccss.conf.sample: Merged revisions 280549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280549 | russell | 2010-07-29 15:35:30 -0500 (Thu, 29 Jul 2010) | 5 lines Add header to ccss.conf to appease oej. (closes issue #17755) Reported by: oej ........ 2010-07-29 19:48 +0000 [r280520] Sean Bright * /, channels/sig_pri.c: Merged revisions 280519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul 2010) | 7 lines Fix compilation error in chan_dahdi (strdupa -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches: strdupa_oops.diff uploaded by malcolmd (license 924) ........ 2010-07-29 19:35 +0000 [r280459-280518] David Vossel * channels/chan_sip.c: respond with 481 when request requiring totag has no totag to match against * main/channel.c, /: Merged revisions 280450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280450 | dvossel | 2010-07-29 14:13:27 -0500 (Thu, 29 Jul 2010) | 25 lines Merged revisions 280449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ ................ ................ 2010-07-29 18:51 +0000 [r280447] Paul Belanger * /, tests/test_utils.c: Merged revisions 280446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280446 | pabelanger | 2010-07-29 14:37:32 -0400 (Thu, 29 Jul 2010) | 2 lines Remove res_crypto dependency. ........ 2010-07-29 16:47 +0000 [r280416] Jean Galarneau * /, apps/app_meetme.c: Merged revisions 280346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280346 | jeang | 2010-07-29 11:07:16 -0500 (Thu, 29 Jul 2010) | 17 lines Merged revisions 280345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ ................ ................ 2010-07-29 16:45 +0000 [r280415] Paul Belanger * /, tests/test_utils.c: Merged revisions 280414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280414 | pabelanger | 2010-07-29 12:44:22 -0400 (Thu, 29 Jul 2010) | 2 lines crypto_loaded_test depends on res_crypto, else test will fail. ........ 2010-07-29 16:26 +0000 [r280395] Russell Bryant * main/rtp_engine.c, /: Merged revisions 280391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280391 | russell | 2010-07-29 11:25:43 -0500 (Thu, 29 Jul 2010) | 2 lines Don't blow up if get_codec() was not provided in the RTP glue. ........ 2010-07-29 15:58 +0000 [r280308-280344] Matthew Nicholson * channels/chan_usbradio.c, /: Merged revisions 280343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280343 | mnicholson | 2010-07-29 10:57:57 -0500 (Thu, 29 Jul 2010) | 4 lines Use PRIx64 instead of PRId64 in format string. related to r280302 ........ * channels/chan_usbradio.c, /: Merged revisions 280302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280302 | pabelanger | 2010-07-28 19:45:34 -0500 (Wed, 28 Jul 2010) | 2 lines Use PRId64 with format_t ........ * channels/chan_usbradio.c: Make chan_usbradio.c build on 64bit platforms. * main/channel.c, channels/chan_local.c, /: Merged revisions 280307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. ................ 2010-07-28 20:50 +0000 [r280270] Jeff Peeler * /, channels/sip/reqresp_parser.c: Merged revisions 280269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280269 | jpeeler | 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010) | 2 lines Give test category missing leading slash ........ 2010-07-28 20:19 +0000 [r280247] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 280235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280235 | rmudgett | 2010-07-28 15:12:16 -0500 (Wed, 28 Jul 2010) | 9 lines Merged revisions 280229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options. ........ ................ 2010-07-28 20:04 +0000 [r280234] Jason Parker * /, sounds/Makefile: Merged revisions 280233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280233 | qwell | 2010-07-28 15:03:22 -0500 (Wed, 28 Jul 2010) | 13 lines Merged revisions 280231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) | 6 lines Work around some silly behavior on BSD. A non-zero exit from a subshell should make the build fail. (closes issue #17621) ........ ................ 2010-07-28 19:37 +0000 [r280226] Terry Wilson * res/res_rtp_asterisk.c, /: Merged revisions 280225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280225 | twilson | 2010-07-28 12:34:42 -0700 (Wed, 28 Jul 2010) | 3 lines Do rtp/rtcp debugging when it is turned on w/o filtering ........ 2010-07-28 18:25 +0000 [r280196] Jason Parker * /, sounds/Makefile: Merged revisions 280195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280195 | qwell | 2010-07-28 13:24:29 -0500 (Wed, 28 Jul 2010) | 16 lines Merged revisions 280193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) | 9 lines Remove unnecessary subshells. Attempt to make checksumming work. Also improves readability. (issue #17621) Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ ........ ................ 2010-07-28 16:53 +0000 [r280162] Sean Bright * /, apps/app_queue.c: Merged revisions 280161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280161 | seanbright | 2010-07-28 12:52:12 -0400 (Wed, 28 Jul 2010) | 15 lines Merged revisions 280160 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines Plug a reference leak in app_queue when adding members dynamically. (closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright ........ ................ 2010-07-28 14:14 +0000 [r280093] Olle Johansson * channels/chan_sip.c: Formatting changes 2010-07-28 13:53 +0000 [r280091] Leif Madsen * contrib/scripts/live_ast, /: Merged revisions 280090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280090 | lmadsen | 2010-07-28 08:52:50 -0500 (Wed, 28 Jul 2010) | 16 lines Merged revisions 280089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500 (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line Update help text to be less confusing. ........ ................ ................ 2010-07-28 13:02 +0000 [r280059] Russell Bryant * /, res/res_crypto.c: Merged revisions 280058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280058 | russell | 2010-07-28 08:01:15 -0500 (Wed, 28 Jul 2010) | 2 lines s/init keys/keys init/ ........ 2010-07-28 01:39 +0000 [r280024] Paul Belanger * channels/chan_usbradio.c, /: Merged revisions 280023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280023 | pabelanger | 2010-07-27 21:37:10 -0400 (Tue, 27 Jul 2010) | 5 lines Resolve compiler warning about formatting (closes issue #17732) Reported by: pabelanger ........ 2010-07-27 21:16 +0000 [r279954] Russell Bryant * /, utils, codecs, main/db1-ast/mpool, Makefile.rules, cdr, formats, codecs/gsm/src, bridges, codecs/lpc10, configure, main/editline, channels/sip, pbx, res/ael, channels, main/stdtime, main/editline/np, main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db, res/ais, res/snmp, funcs, main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, makeopts.in, res: Merged revisions 279953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010) | 5 lines Add --enable-coverage option to configure script. This option enables the proper compiler flags for tracking code coverage, which is useful along side automated testing. ........ 2010-07-27 20:59 +0000 [r279951] David Vossel * main/channel.c, /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 279949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500 (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ ................ ................ 2010-07-27 19:55 +0000 [r279917] Russell Bryant * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 279916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines Fix inband DTMF detection on outgoing ISDN calls. This is a regression from the sig_pri split from chan_dahdi. When a call is first initiated, the inband DTMF detector is not enabled if it's an outgoing ISDN call. However, it needs to be turned on once the media path starts up. This handling was put back in the open_media() callback of chan_dahdi. In sig_pri, open_media() calls were added to a few places where it was needed, including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the patch! ........ 2010-07-27 18:55 +0000 [r279888] Mark Michelson * /, channels/chan_sip.c: Merged revisions 279887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul 2010) | 16 lines Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address. In this particular scenario, there was no value from parsing the port out, so I just removed that logic. And while I was messing around in the function, I changed some variable names to be more descriptive. (closes issue #17661) Reported by: oej Patches: 17661.diff uploaded by mmichelson (license 60) ........ 2010-07-27 16:41 +0000 [r279851] Jason Parker * /, sounds/Makefile: Merged revisions 279850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279850 | qwell | 2010-07-27 11:40:05 -0500 (Tue, 27 Jul 2010) | 9 lines Merged revisions 279849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) | 1 line Simply sounds/Makefile some more. ........ ................ 2010-07-27 16:11 +0000 [r279818] David Vossel * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 279817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) | 2 lines fix sip transaction match with authentication, fix confusing log message when using getaddrinfo ........ 2010-07-27 16:08 +0000 [r279816] Russell Bryant * main/channel.c, channels/chan_dahdi.c, /: Merged revisions 279636,279815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) | 2 lines Ignore a control subclass of -1 in ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in addition to "channel" in chan_dahdi.conf. Review: https://reviewboard.asterisk.org/r/804 ........ 2010-07-27 15:16 +0000 [r279786] Mark Michelson * /, channels/chan_sip.c: Merged revisions 279785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279785 | mmichelson | 2010-07-27 10:15:22 -0500 (Tue, 27 Jul 2010) | 20 lines Merged revisions 279784 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines Fix bad behavior of dynamic_exclude_static option in sip.conf. We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD ........ ................ 2010-07-27 03:02 +0000 [r279727-279756] Paul Belanger * channels/chan_dahdi.c, /: Merged revisions 279755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, 26 Jul 2010) | 10 lines If dringXcontext is null, fallback to default context value. (closes issue #17693) Reported by: iasgoscouk Patches: issue17693.patch uploaded by pabelanger (license 224) Tested by: iasgoscouk Review: https://reviewboard.asterisk.org/r/803/ ........ * /, main/http.c: Merged revisions 279726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul 2010) | 9 lines Use ast_sockaddr_setnull() when http is not enabled. Otherwise, ast_tcptls_server_start() will still start http. (closes issue #17708) Reported by: pabelanger Patches: http.patch uploaded by pabelanger (license 224) ........ 2010-07-27 01:39 +0000 [r279725] Russell Bryant * CHANGES: Make a formatting change. (Demonstrating the commit IRC bot to pabelanger) 2010-07-26 23:35 +0000 [r279692] Paul Belanger * /, CHANGES, UPGRADE-1.8.txt: Merged revisions 279689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines Updated documentation for FAX logger level. ........ 2010-07-26 23:06 +0000 [r279659] Jason Parker * /, sounds/Makefile.380 (removed), configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381 (removed), configure.ac, sounds/Makefile (added): Merged revisions 279658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279658 | qwell | 2010-07-26 18:03:38 -0500 (Mon, 26 Jul 2010) | 12 lines Merged revisions 279657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines Really fix sounds Makefile (and make it readableish). There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. ........ ................ 2010-07-26 21:21 +0000 [r279602-279624] Tilghman Lesher * /, configure, configure.ac: Merged revisions 279619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279619 | tilghman | 2010-07-26 16:20:12 -0500 (Mon, 26 Jul 2010) | 9 lines Merged revisions 279609 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26 Jul 2010) | 2 lines Dunno why this worked on my machine, but it works better this way. ........ ................ * /, res/res_config_ldap.c: Merged revisions 279601 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279601 | tilghman | 2010-07-26 16:07:45 -0500 (Mon, 26 Jul 2010) | 19 lines Merged revisions 279597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | 13 lines Apply all patches in: https://issues.asterisk.org/view.php?id=13573 (closes issue #13573) Reported by: navkumar Patches: res_config_ldap-category.diff uploaded by navkumar (license 580) res_config_ldap.patch uploaded by bencer (license 961) res_config_ldap uploaded by bencer (license 961) Tested by: suretec ........ ................ 2010-07-26 21:07 +0000 [r279600] Gavin Henry * /: Merged revisions 279598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279598 | ghenry | 2010-07-26 21:58:12 +0100 (Mon, 26 Jul 2010) | 21 lines Merged revisions 279597 via svnmerge from https://origsvn.digium.com/svn/asterisk/1.6.2 ----------------------------------------------------------------------- r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) | 13 lines Apply all patches in: https://issues.asterisk.org/view.php?id=13573 [^] (closes issue 0013573) Reported by: navkumar Patches: res_config_ldap-category.diff uploaded by navkumar (license 580) res_config_ldap.patch uploaded by bencer (license 961) res_config_ldap uploaded by bencer (license 961) Tested by: suretec ------------------------------------------------------------------------ ........ 2010-07-26 20:00 +0000 [r279569] David Vossel * /, channels/chan_sip.c, channels/sip/reqresp_parser.c, channels/sip/include/sip.h, channels/sip/include/reqresp_parser.h: Merged revisions 279568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) | 21 lines transaction matching using top most Via header This patch modifies the way chan_sip.c does transaction to dialog matching. Asterisk now stores information in the top most Via header of the initial incoming request and compares that against other Requests that have the same call-id. This results in Asterisk being able to detect a forked call in which it has received multiple legs of the fork. I completely stripped out the previous matching code and made the comparisons a little more explicit and easier to understand. My comments in the code should offer all the details involving this patch. This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to find multiple dialogs with the same call-id. Since the callback function was returning (CMP_MATCH | CMP_STOP) only the first item found was being returned. I fixed this by making a new callback function for finding multiple dialogs that only returns (CMP_MATCH) on a match allowing for multiple items to be returned. Review: https://reviewboard.asterisk.org/r/776/ ........ 2010-07-26 19:58 +0000 [r279567] Paul Belanger * /, CHANGES, UPGRADE-1.8.txt, configs/logger.conf.sample: Merged revisions 279566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) | 8 lines Add documentation for FAX logger level. (closes issue #17715) Reported by: vrban Patches: 17715.patch uploaded by pabelanger (license 224) Tested by: vrban ........ 2010-07-26 19:20 +0000 [r279564] Tilghman Lesher * /, sounds/Makefile.380 (added), configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381 (added), configure.ac, sounds/Makefile (removed): Merged revisions 279562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279562 | tilghman | 2010-07-26 14:18:26 -0500 (Mon, 26 Jul 2010) | 9 lines Merged revisions 279561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) | 2 lines Use a special Makefile for noobs who still have GNU Make 3.80. ........ ................ 2010-07-26 16:44 +0000 [r279533] Mark Michelson * /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sip/reqresp_parser.c: Merged revisions 279504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) | 14 lines Allow for systems without locale support to be usable. A recent change to SIP URI comparison code added a locale-specific string comparison to the mix, and certain systems do not support such functions. This fix allows for those systems to still use Asterisk 1.8 (closes issue #17697) Reported by: pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347) Tested by: mmichelson ........ 2010-07-26 15:44 +0000 [r279503] Sean Bright * /, autoconf/ast_ext_lib.m4: Merged revisions 279502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279502 | seanbright | 2010-07-26 11:43:54 -0400 (Mon, 26 Jul 2010) | 12 lines Merged revisions 279501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, 26 Jul 2010) | 5 lines Expand the correct value within AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth ........ ................ 2010-07-26 03:28 +0000 [r279473] Tilghman Lesher * /, formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c, formats/format_wav_gsm.c, formats/format_sln16.c, formats/format_siren7.c, formats/format_ilbc.c, formats/format_vox.c, formats/format_pcm.c, formats/format_h263.c, formats/format_g723.c, formats/format_h264.c, formats/format_siren14.c, formats/format_jpeg.c, formats/format_g726.c, formats/format_gsm.c, formats/format_g719.c, formats/format_g729.c: Merged revisions 279472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279472 | tilghman | 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010) | 2 lines Formats need to load before apps, because some apps call ast_format_str_reduce() at load time. ........ 2010-07-25 21:28 +0000 [r279443] Paul Belanger * /, tests/test_func_file.c: Merged revisions 279442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279442 | pabelanger | 2010-07-25 17:26:42 -0400 (Sun, 25 Jul 2010) | 2 lines Add trailing backslash to silence warning message. ........ 2010-07-25 18:22 +0000 [r279391-279413] Tilghman Lesher * /, cdr/cdr_odbc.c: Merged revisions 279410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279410 | tilghman | 2010-07-25 13:21:27 -0500 (Sun, 25 Jul 2010) | 8 lines Don't re-register CDR module on reload. (closes issue #17304) Reported by: jnemeth Patches: 20100507__issue17304.diff.txt uploaded by tilghman (license 14) Tested by: jnemeth ........ * /, main/logger.c: Merged revisions 279390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279390 | tilghman | 2010-07-25 12:32:21 -0500 (Sun, 25 Jul 2010) | 8 lines Don't assume qlog is open. (closes issue #17704) Reported by: vrban Patches: issue17704.patch uploaded by pabelanger (license 224) Tested by: vrban ........ 2010-07-24 20:49 +0000 [r279274-279315] Paul Belanger * Makefile, /: Merged revisions 279314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279314 | pabelanger | 2010-07-24 16:47:52 -0400 (Sat, 24 Jul 2010) | 7 lines Remove duplicate -c flag when using $(INSTALL) (closes issue #17695) Reported by: pabelanger Patches: Makefile.diff uploaded by pabelanger (license 224) ........ * /, include/asterisk/netsock2.h: Merged revisions 279280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279280 | pabelanger | 2010-07-24 14:18:43 -0400 (Sat, 24 Jul 2010) | 8 lines Check if ast_sockaddr is NULL then return. (closes issue #17677) Reported by: outcast Patches: issue0017677.patch uploaded by pabelanger (license 224) Tested by: elguero ........ * main/manager.c, /: Merged revisions 279273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul 2010) | 6 lines Default sin_family to AF_INET for TCP / TLS Bindaddress. Otherwise, 'manager show settings' will generate errors if manager is not enabled. ........ 2010-07-23 22:24 +0000 [r279156-279245] Richard Mudgett * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 279227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279227 | rmudgett | 2010-07-23 17:20:47 -0500 (Fri, 23 Jul 2010) | 21 lines Merged revisions 279207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ ................ ................ * channels/chan_dahdi.c: Make "dahdi show channels" show an outgoing called number. The "dahdi show channels" extension column previously only showed the called number of an incoming call. It now shows the called number for an outgoing call as well. (closes issue #17653) Reported by: amazinzay Patches: issue17653_trunk.txt uploaded by rmudgett (license 664) 2010-07-23 19:17 +0000 [r279116-279118] Russell Bryant * UPGRADE.txt, UPGRADE-1.8.txt (added): Shuffle UPGRADE.txt files for 1.10. * CHANGES: Start a new section in CHANGES for 1.10. 2010-07-23 18:56 +0000 [r279115] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 279113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279113 | tilghman | 2010-07-23 13:56:04 -0500 (Fri, 23 Jul 2010) | 2 lines Silly 64-bit compilers (who uses 64-bit anyway?) ........ 2010-07-23 18:22 +0000 [r279063-279084] Russell Bryant * /: Remove old properties. * /: Add branch-1.8-merged and branch-1.8-blocked properties to trunk.