2007-04-24 Russell Bryant * Asterisk 1.4.3 released. 2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant * main/manager.c, /: Merged revisions 61786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | 4 lines Don't crash if a manager connection provides a username that exists in manager.conf but does not have a password, and also requests MD5 authentication. (ASA-2007-012) ........ * main/channel.c, include/asterisk/channel.h: Improve DTMF handling in ast_read() even more in response to a discussion on the asterisk-dev mailing list. I changed the enforced minimum length of a digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in between digits. These values are not configurable in a configuration file right now, but they can be easily changed near the top of main/channel.c. 2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor() ........ 2007-04-24 16:16 +0000 [r61774] Russell Bryant * main/dial.c: Add a few more state changes in handle_frame_ownerless() so that the SLA code will get notified of these changes even when an owner channel is not provided. This isn't from a specific bug report, it's just something I noticed while poking around. 2007-04-24 16:07 +0000 [r61772] Joshua Colp * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford) ........ 2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant * main/pbx.c: Some dialplan functions, such as CUT(), expect to operate on variables on a channel. So, this little hack lets them work in places where a channel doesn't exist, such as within DUNDi configuration. (issue #9465, reported and patched by Corydon76, testing by blitzrage) * main/channel.c: Ensure that digits passing through Asterisk have a reasonable minimum length. It is currently 100 ms. If someone thinks this should be different, feel free to speak up. (related to issues #8944, #9250, and #9348) 2007-04-20 21:35 +0000 [r61705-61707] Jason Parker * main/rtp.c: Avoid invalid seqno cycling detection. Per comment from Dave Troy: This adds back in some simple typecasting I had in an earlier version which I realize now may be breaking things. Issue #9554. * main/loader.c, /: Merged revisions 61704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 lines Fix an issue that I noticed while looking over issue 9571. The reload timestamp was getting set after reloading the built-in stuff, and before the modules. ........ 2007-04-20 20:42 +0000 [r61697] Russell Bryant * main/rtp.c: Remove a stray debug message introduced by a recent commit. 2007-04-20 19:51 +0000 [r61694] Jason Parker * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 lines If the '* to hangup' option is not enabled, we don't need to disable * as a valid exit key. If it was enabled, this statement would've never been checked in the first place. Issue #9552 ........ 2007-04-20 18:19 +0000 [r61690] Russell Bryant * main/config.c, apps/app_voicemail.c, main/manager.c, include/asterisk/config.h: Fix the UpdateConfig manager action to properly treat "variables" and "objects" differently (a=b versus a=>b). (issue #9568, reported by pari, patch by me) 2007-04-19 08:37 +0000 [r61686] Olle Johansson * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........ 2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher * main/manager.c: Bug 9557 - simple reason why reading a function always returned NULL * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c, funcs/func_groupcount.c, /, funcs/func_timeout.c, funcs/func_cdr.c: Merged revisions 61680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) | 5 lines Bug 9557 - Specifying the GetVar AMI action without a Channel parameter can cause Asterisk to crash. The reason this needs to be fixed in the functions instead of in AMI is because Channel can legitimately be NULL, such as when retrieving global variables. ........ 2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming * sounds/Makefile: allow external build systems to extract the required sound file versions 2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson * main/rtp.c: Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin! * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy) 2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson * channels/chan_sip.c: #9483, half of patch by twilson to solve 302 redirect issues * /: Blocking AstHoloPatch from 1.2 2007-04-13 21:17 +0000 [r61658] Steve Murphy * main/cdr.c: This is a fix to the way CDR merge handles the data that results from ForkCDR. 2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140) ........ * apps/app_speech_utils.c: Do not bother looking for a result if none are present. * channels/chan_sip.c: For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg) 2007-04-13 17:10 +0000 [r61645] Russell Bryant * apps/app_voicemail.c: Eliminate a compiler warning with ODBC_STORAGE enabled so that it will build under dev-mode. 2007-04-13 17:01 +0000 [r61644] Steve Murphy * channels/chan_oss.c: A fix for chan_oss that resulted from the CDR changes; it helps to use the right info. 2007-04-13 16:32 +0000 [r61641] Joshua Colp * channels/chan_sip.c: Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa) 2007-04-11 16:05 +0000 [r61477] Russell Bryant * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines If someone sets the "useragent" option in sip.conf to be empty, then don't add the User-Agent header at all. It is an optional header, anyway. Also, the bug report says that some of Japan's SIP providers don't allow it for some weird reason. (issue #9488, reported by makoto, fixed by me) ........ 2007-04-11 15:39 +0000 [r61443] Nadi Sarrar * channels/chan_misdn.c: Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk.. 2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines Fix a bug with switching between host=dynamic and using specific hosts for peers. The code would only reset the peer's address when it is dynamic if it was a new peer structure. Now, it will also reset the address if it was already in the peer list, but before the reload, it was not dynamic. (issue #9515, reported by caio1982, fixed by me) ........ * main/http.c: Add "svgz" to the mimetypes table. (issue #9510, bkruse) In passing, constify the elements of the mimetypes table. * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines Remove the attempt at reporting configuration errors in sip.conf. This can cause a bunch of improper messages when using realtime. I give up. As oej tried to convince me when I put this in, there is just no easy way to do it. (inspired by a message on the -dev list) ........ 2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar * channels/chan_misdn.c: Export AOCD variables on misdn_hangup. * channels/chan_misdn.c: Ignore facility messages in case we don't have a corresponding channel object. * channels/chan_misdn.c: AOCD's are now exported to asterisk channel variables. 2007-04-10 16:05 +0000 [r61220] Russell Bryant * main/Makefile, main/http.c, main/minimime (removed): File upload support was added to solve some needs for the Asterisk GUI. However, after much discussion, it has been decided that adding this to 1.4 is not in the best interests of the project. It has been removed here, but will remain in trunk. 2007-04-10 12:43 +0000 [r61183] Nadi Sarrar * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines msns config parameter defaults to '*' ........ 2007-04-10 05:18 +0000 [r61136] Steve Murphy * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a previous fix to overcome a compiler warning; the app NoCDR() has been updated to mark the channel CDR as POST_DISABLED instead of destroying the CDR; this way its flags are propagated thru a bridge and the CDR is actually dropped. The cases where only one channel in a bridge has a CDR was cleaned up. 2007-04-09 19:58 +0000 [r61072] Olle Johansson * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines - Don't send ActionID before Response: header. - Don't use a blank in an AMI header ........ 2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming * main/minimime/mm_envelope.c, res/res_features.c: fix up some warnings found using --enable-dev-mode * main/minimime/Doxyfile (removed), main/minimime/tests/messages/CVS (removed), main/minimime/tests/CVS (removed): remove some more stuff we don't need 2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant * main/minimime/test (removed): Remove another directory that should no longer be there * main/minimime/Make.conf (removed), main/minimime/mytest_files (removed), main/minimime/.cvsignore (removed), main/minimime/sys (removed), main/minimime/mm-docs (removed): Remove various files that I thought I already removed. 2007-04-09 19:05 +0000 [r61022] Jason Parker * apps/app_queue.c: Use the appropriate interface name with COMPLETECALLER. Issue 9395. 2007-04-09 18:32 +0000 [r60989] Steve Murphy * channels/chan_oss.c, main/channel.c, main/cdr.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c: This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. 2007-04-09 18:02 +0000 [r60984] Olle Johansson * res/res_jabber.c: Add final new line after JabberEvent 2007-04-09 17:22 +0000 [r60936] Jason Parker * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 lines Allow matching on names shorter than 3 chars. This also fixes the case where somebody wants to match on less then 3 chars. Issue 9071 ........ 2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list). ........ * channels/chan_local.c, /: Merged revisions 60846 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid. ........ 2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 lines When calling a device that then forwards us elsewhere... we have to make our channels compatible if it is the only channel being dialed. (issue #9445 reported by marcelbarbulescu) ........ * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy) 2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007) | 2 lines Gosub called within a Macro resets the arguments improperly and causes general weirdness. (Issue 8329) ........ * main/http.c: Fix --enable-dev-mode * channels/chan_oss.c: Off by one error, resulting in a crash (Issue 9500) * /, main/file.c: Merged revisions 60660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) | 2 lines Bug 9486 - memory leak when opening a filestream ........ 2007-04-06 20:58 +0000 [r60603] Russell Bryant * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile, main/minimime/mimeparser.yy.c, main/minimime/minimime.c, main/manager.c, main/minimime/mm_mimepart.c, main/minimime/test.sh, configure, include/asterisk/compat.h, main/strcompat.c, main/minimime/mm_internal.h, main/http.c, main/minimime/tests/parse.c, main/minimime/mm_base64.c, main/minimime/mm_mimeutil.c, main/minimime/mm.h, main/minimime/tests, main/minimime/mm_header.c, main/minimime/mm_error.c, main/Makefile, main/minimime/mm_codecs.c, main/minimime/mm_param.c, configure.ac, main/minimime/Makefile, main/minimime/mm_init.c, include/asterisk/manager.h, main/minimime/strlcpy.c, configs/http.conf.sample, main/minimime/mm_parse.c, main/minimime/tests/create.c, main/minimime/mm_contenttype.c, main/minimime/mm_util.c, main/minimime/mm_envelope.c, main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c, main/minimime/tests/messages/test2.txt, main/minimime/tests/messages/test3.txt, main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c, main/minimime/tests/messages/test4.txt, main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h, main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c, main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt, main/minimime/mimeparser.l, main/minimime/mm_context.c, main/minimime/mimeparser.tab.h, main/minimime (added), main/minimime/mm_warnings.c, main/minimime/mm_queue.h, main/minimime/tests/messages, include/asterisk/autoconfig.h.in, main/minimime/mimeparser.y, Makefile.moddir_rules, main/minimime/sys, main/minimime/tests/Makefile: To be able to achieve the things that we would like to achieve with the Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. 2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard * UPGRADE.txt: clarified a sentence in the format_wav section * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and plan to remove GAIN code from trunk 2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant * apps/app_meetme.c: When a station picks up a trunk that was on hold, make the hints reflect that nobody has the trunk on hold anymore. * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459, reported by francesco_r, fixed by me) * The original behavior was that if one station put a call on hold, another one picked it up, and then hung up, the code would still consider the call on hold by the first station, so the trunk would not be hung up. However, to better comply with what most people seem to expect it to behave, it will now hang up the trunk. * Fix a problem with "barge=no". This was only intended to prevent people from joining calls that are in progress. However, it also prevented other people from picking up a call that was on hold. This has been fixed. * When there are no active stations on a trunk and it is on hold, the code now indicates the HOLD and UNHOLD conditions to the trunk channel. This allows music on hold to be played to the trunk when it is on hold. 2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson * channels/chan_zap.c: Make sure we check the faxdetect option before doing fax processing * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines There should only be one code path for doing DTMF conditionals on channels. This fixes it. ........ 2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) | 2 lines remove undocumented 'cardsmode' parameter and stop searching for transcoders during reload() ........ 2007-04-06 01:14 +0000 [r60361] Joshua Colp * res/res_speech.c, apps/app_speech_utils.c, include/asterisk/speech.h: Add support for returning different types of results (ie: NBest). 2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard * formats/format_wav.c: modified default GAIN for issue 5823, thanks jrwalliker 2007-04-05 22:35 +0000 [r60323] Steve Murphy * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes. 2007-04-05 16:10 +0000 [r60268] Jason Parker * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 lines Just because we can't find the voicemail configuration file, doesn't mean that the module failed to load. The user could be using realtime. Issue #9473 ........ 2007-04-05 15:47 +0000 [r60265] Russell Bryant * main/http.c: Add the MIME type for gif by request from Pari 2007-04-05 12:55 +0000 [r60214] Joshua Colp * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa) ........ 2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant * main/manager.c, /: Merged revisions 60134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | 6 lines It is valid to redirect channels via the manager interface that are not in the UP state. Instead of checking for that to prevent to ensure a dead channel doesn't get redirected, just use the ast_check_hangup() API call. (issue #9457, reported by Callmewind, patch by me) (related to issue #8977) ........ * channels/chan_sip.c: Add a Content-Length of 0 to the response built by transmit_response_with_unsupported(). (issue #9454, reported by makoto, fixed by me) * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines Fix the return value of handle_common_options() so that it always properly indicates whether it handled the option or not. (issue #9455, reported by Netview, fixed by me) ........ * apps/app_meetme.c: Fix a problem where if a trunk was hung up while it was on hold, all of the hints would reflect the line still on hold, even though it should reflect that it is back to not in use. (issue #9459, reported by francesco_r, fixed by me) * /: Blocked revisions 60016 via svnmerge ........ r60016 | russell | 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a missing "\r\n" in the body of the NOTIFY that is sent to indicate the status of a transfer. (issue #9388, reported by rarritt) ........ * /: Blocked revisions 60014 via svnmerge ........ r60014 | russell | 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the more generic check for "sed -r" support that was already present in 1.4. (related to issue #9399) ........ * /: Blocked revisions 60012 via svnmerge ........ r60012 | russell | 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On Darwin, the -r argument to sed is not valid. It has to be -E. (issue #9399, reported by jcovert) ........ 2007-04-03 19:40 +0000 [r59963] Joshua Colp * apps/app_speech_utils.c: Don't clash when a person both speaks and uses DTMF. 2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines Don't attempt to report configuration errors in build_user(). oej pointed out that for a "friend" entry, this won't work, because all user options are valid for peers, but not the other way around. ........ * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines Make chan_sip report when it encounters an unknown option. (issue #9440, reported by nightcrawler) ........ * /, main/app.c: Merged revisions 59886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | 5 lines When doing a built-in blind or attended transfer, restore the ability to use '#' to terminate the number and immediately do the transfer instead of having to dial the number and just wait for the feature digit timeout. (issue #8366, xueliangliang) ........ * Makefile: Ensure that menuselect gets executed in dependency check mode every time you run make. 2007-04-03 11:02 +0000 [r59804] Nadi Sarrar * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h: Merged revisions 59788,59803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines Use the new sysfs way of mISDN 1.2 to check if a port is NT or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ 2007-04-03 07:20 +0000 [r59774] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: Merged revisions 59623-59624,59639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line we can now make 30 channels on a PRI (before we forgot chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour ........ 2007-04-02 18:58 +0000 [r59724] Joshua Colp * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 lines Increase the maximum size for a string of mailboxes to 1024. (issue #9270 reported by rtucker) ........ 2007-04-02 17:31 +0000 [r59688] Steve Murphy * pbx/pbx_ael.c: continue in for-loop should go to the incrementer, not the test. As per 9435, thanks to marcelbarbulescu 2007-04-02 15:39 +0000 [r59654] Russell Bryant * main/netsock.c, /: Merged revisions 59608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock. This is needed by the patch that went in for issue 7874. chan_iax2 needs to be able to create socket that is lisetning on INADDR_ANY, but also be able to bind sockets to specific addresses. (Thanks to Stevenson on the asterisk-dev mailing list for explaining why this flag was needed.) ........ 2007-03-30 22:50 +0000 [r59573] Jason Parker * configure, main/Makefile, acinclude.m4: Add linux-uclibc host arch..."thingy". Sorry, I don't know what it's called... 2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, include/asterisk/cdr.h: several changes via kpflemings review * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, include/asterisk/cdr.h: These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. * configs/extensions.conf.sample: A small clarification to keep bugs from being filed, and confusion from rising, if clearglobalvars is set, and globals are set in the AEL file. (9419) 2007-03-29 17:43 +0000 [r59363] Russell Bryant * res/res_jabber.c: When building a response to a subscription, the "from" must be the full Jabber ID. This fixes some problems where jabber users are not able to add their Asterisk account to their user list, since they are unable to get Asterisk to approve their subscription. (issue #8210, reported by caspy, and verified by bradtem) 2007-03-29 17:38 +0000 [r59361] Joshua Colp * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 lines Keep a global array of variables indicating whether certain conference rooms are in use. This ensures that two people going into a new dynamic conference when the 'e' option is set don't go into the same conference room. (issue #8835 reported by eliel) ........ 2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant * main/rtp.c, /: Merged revisions 59357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines If an error occurs when reading from an RTP socket, and the error code does not indicate that we should try again, then return NULL instead of a "null frame". This will prevent Asterisk from trying over and over again, and eventually causing the system to crash. (issue #8285, john) ........ * /: Blocked revisions 59355 via svnmerge ........ r59355 | russell | 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport the change to chan_iax2 to return NULL instead of a "null frame" from its read callback. See revision 59341 to the 1.4 branch for more info. ........ * channels/chan_iax2.c: When the IAX2 read callback gets called, return NULL instead of a "null frame". This will cause Asterisk to hangup the call instead of keep trying whatever it was doing. Under normal conditions, this function would *never* be called. However, the author of this patch says an error will occur that will cause it to get called every 100 thousand calls or so. When this does happen, it puts the channel in a loop that eventually brings down the system. So, hangup up the call is certainly a better alternative. (issue #8286, john) * Makefile: Export the GTK2 library and include information to sub Makefiles. 2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007) | 3 lines Issue 9415 - No point to getting a diagnostic field if we aren't doing anything with the information. (Plus, it tends to crash the Postgres ODBC driver.) ........ * /: Blocked revisions 59299 via svnmerge ........ r59299 | tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines Change ENV section to use setenv, instead of putenv (Alexandru Pirvulescu , reported via -dev list) ........ 2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher * res/res_odbc.c: Another crash that I thought we had fixed already - Issue 9396 * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007) | 2 lines Oops ........ * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007) | 2 lines Fix a few remaining bad mmap(2) return values ........ 2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) | 3 lines Fix the check of the return value from mmap(). Thanks to Corydon for catching this one. ........ * apps/app_directory.c: Fix app_directory to actually compile with ODBC_STORAGE, and update the code to the latest res_odbc API. * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used. The Makefile did not properly ensure that this information got copied from what was selected for app_voicemail. (issue #9224) * channels/chan_sip.c: Fix the check that ensures that the CHANNEL function's first argument is "rtpqos". Thanks, Corydon. :) 2007-03-27 18:16 +0000 [r59261] Steve Murphy * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes asterisk), kpfleming pointed on asterisk-dev, that DECLINE in this case the proper thing to do. This change now has it doing the proper thing. 2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0 instead of having each interface explicitly listed. (issue #7874, patch by stevens) ........ * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS function to just be additional parameter of the CHANNEL function. This way, it will be possible for other RTP based channel drivers to expose this information in the future. 2007-03-27 15:00 +0000 [r59254] Christian Richter * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line fixed #9355 ........ 2007-03-26 21:45 +0000 [r59230] Tilghman Lesher * channels/chan_sip.c: Oops, this should be case insensitive 2007-03-26 21:41 +0000 [r59228] Steve Murphy * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes asterisk). I turned a duplicate context from a WARNING to an ERROR. Now you get a module load failure, and asterisk just exits. That's better than a crash, right\? 2007-03-26 21:37 +0000 [r59227] Tilghman Lesher * channels/chan_sip.c: Change this to a single dp function to make oej happy. 2007-03-26 20:06 +0000 [r59225] Steve Murphy * main/config.c: Fix for 9257; by eliminating the globals in main/config.c, we make it thread-safe, which is a minimum requirement. 2007-03-26 19:34 +0000 [r59223] Joshua Colp * apps/app_speech_utils.c: Add ability to specify no timeout. This means as soon as the prompt is done playing it moves on to the next priority. 2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant * apps/app_voicemail.c: Somehow the code for building the email for voicemail got out of sync. This change makes a few tweaks to get 1.4 in sync with trunk. (issue #9301) * apps/app_meetme.c: Fix some codec negotiation problems when CallerID support is not enabled in SLA. (issue #9308, reported by twilson) 2007-03-26 18:13 +0000 [r59213] Joshua Colp * apps/app_speech_utils.c: Make SpeechBackground obey the digit timeout value. 2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant * channels/chan_sip.c: Rename the new dialplan functions to match the variable name * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) 2007-03-26 17:38 +0000 [r59206] Steve Murphy * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE, and STANDALONE_AEL 2007-03-26 15:25 +0000 [r59202] Nadi Sarrar * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure, include/asterisk/autoconfig.h.in, channels/misdn/Makefile, channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it. * add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in' (the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected). 2007-03-26 15:16 +0000 [r59200] Joshua Colp * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the aelparse binary! DONT_OPTIMIZE should now work once again. 2007-03-24 01:39 +0000 [r59195] Joshua Colp * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines Only try to handle a response if it has a response code. (ASA-2007-011) ........ 2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy * /: blocking out the fix in 59187... already incorporated here * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 line Added a few words in the Macro doc strings about the behavior of macros with hangups (et al.), as per 9337 ........ 2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming * channels/chan_sip.c: don't allow string input to overrun the buffer to hold it (ASA-2007-010) * channels/chan_misdn.c: remove variables that are no longer used (--enable-dev-mode is good, developers should be using it) 2007-03-22 14:40 +0000 [r59145] Steve Murphy * utils/Makefile: The stuff in utils was compiling with -O6 even if DONT_OPTIMIZE is set in menuconfig. Added the include to fix that 2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp * main/http.c: Add svg mimetype for pari. * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 lines Indicate the filename changed when it is changed. (issue #9311 reported by jsmith) ........ * channels/chan_sip.c: Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown) 2007-03-20 21:04 +0000 [r59078] Tilghman Lesher * main/logger.c: Fix defines for inline stack backtraces (only used by developers anyway) 2007-03-20 20:42 +0000 [r59076] Joshua Colp * channels/iax2-parser.c: Copy len variable as well, should fix remaining IAX2 DTMF issues. 2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should return it to its previous, untouched, state. * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h: The fix for the AEL <> (bug 9316) is here... 2007-03-20 13:16 +0000 [r59064] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: Merged revisions 58849-58850,59062-59063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line avoid sending a disconnect when we already received one. ........ r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line modified a loglevel ........ 2007-03-19 Jason Parker * Asterisk 1.4.2 released. 2007-03-19 22:29 +0000 [r59049] Tilghman Lesher * funcs/func_strings.c: Oops, this should have been a %d all along 2007-03-19 15:52 +0000 [r59042] Joshua Colp * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295 reported by ajohnson) 2007-03-19 15:42 +0000 [r59040] Tilghman Lesher * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported via -dev list) 2007-03-18 20:37 +0000 [r59037] Olle Johansson * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) 2007-03-18 16:36 +0000 [r59035] BJ Weschke * apps/app_followme.c: Don't return a non-zero return code if the profile doesn't exist, to match what the documentation says it already does. (#9307 Reported by kkiely) 2007-03-16 16:12 +0000 [r58992] Joshua Colp * apps/app_page.c: Wait for the async thread to exit when hanging up all of the paged phones under all circumstances. (issue #9181 reported by PhilSmith) 2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant * configs/sla.conf.sample: fix a couple SLA documentation references * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added), doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added), doc/channelvariables.txt (added), doc/ael.txt (added), doc/billing.tex (removed), build_tools/prep_tarball, doc/callingpres.txt (added), doc/enum.txt (added), doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added), doc/cdrdriver.tex (removed), build_tools/make_buildopts_h, doc/security.txt (added), doc/imapstorage.txt (added), doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed), doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac, doc/iax.txt (added), doc/ael.tex (removed), doc/channelvariables.tex (removed), doc/enum.tex (removed), doc/security.tex (removed), doc/math.txt (added), Makefile, doc/imapstorage.tex (removed), doc/privacy.tex (removed), doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt (added), apps/app_voicemail.c, doc/cliprompt.txt (added), doc/chaniax.txt (added), doc/app-sms.txt (added), doc/ast_appdocs.tex (removed), doc/realtime.tex (removed), doc/ices.txt (added), doc/dundi.tex (removed), doc/linkedlists.txt (added), doc/queuelog.txt (added), doc/extconfig.txt (added), doc/radius.txt (added), doc/cliprompt.tex (removed), doc/chaniax.tex (removed), doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex (removed), doc/ices.tex (removed), doc/asterisk.tex (removed), doc/queuelog.tex (removed), doc/configuration.txt (added), doc/asterisk-conf.txt (added), doc/sla.pdf (added), doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt (added), doc/mp3.tex (removed), doc/configuration.tex (removed), doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added), doc/channels.txt (added), doc/ip-tos.tex (removed), doc/extensions.txt (added), doc/queues-with-callback-members.txt (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added), doc/misdn.txt (added), doc/manager.txt (added), doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), doc/billing.txt (added), doc/localchannel.txt (added), doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt (added), doc/00README.1st (added): Making these documentation changes in the 1.4 branch upset various people, so these chanes will only be done in the trunk. * build_tools/prep_tarball: Add the --pdf option to the usage of rubber in prep_tarball * Makefile, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add configure script checking for GTK2 and some additional Makefile targets to support gmenuselect 2007-03-15 23:52 +0000 [r58946] Tilghman Lesher * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match common syntax and update the resulting appdocs TeX file 2007-03-15 23:24 +0000 [r58941] Russell Bryant * doc/asterisk.tex: add a link to the rubber homepage 2007-03-15 23:11 +0000 [r58939] Tilghman Lesher * apps/app_setcdruserfield.c, main/pbx.c, apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c: Expand deprecation warnings from simply warning on use to the builtin documentation. 2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant * doc/asterisk.tex, Makefile: Add Asterisk version information to the generated PDF * build_tools/prep_tarball: have prep_tarball attempt to build asterisk.pdf 2007-03-15 22:32 +0000 [r58933] Tilghman Lesher * funcs/func_realtime.c: Function works fine, but the documentation is backwards. 2007-03-15 22:25 +0000 [r58931] Russell Bryant * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex (added), doc/freetds.txt (removed), doc/odbcstorage.txt (removed), configure, doc/sla.tex, doc/cygwin.txt (removed), doc/model.txt (removed), doc/channelvariables.txt (removed), doc/ael.txt (removed), doc/billing.tex (added), doc/callingpres.txt (removed), doc/enum.txt (removed), doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed), doc/cdrdriver.tex (added), build_tools/make_buildopts_h, doc/security.txt (removed), doc/imapstorage.txt (removed), doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added), doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac, doc/iax.txt (removed), doc/ael.tex (added), doc/channelvariables.tex (added), doc/enum.tex (added), doc/security.tex (added), doc/math.txt (removed), Makefile, doc/imapstorage.tex (added), doc/privacy.tex (added), doc/realtime.txt (removed), doc/dundi.txt (removed), doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), doc/ast_appdocs.tex (added), doc/realtime.tex (added), doc/ices.txt (removed), doc/dundi.tex (added), doc/linkedlists.txt (removed), doc/queuelog.txt (removed), doc/extconfig.txt (removed), doc/radius.txt (removed), doc/cliprompt.tex (added), doc/chaniax.tex (added), doc/hardware.txt (removed), doc/mp3.txt (removed), doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex (added), doc/queuelog.tex (added), doc/configuration.txt (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added), doc/h323.txt (removed), doc/mp3.tex (added), doc/configuration.tex (added), doc/asterisk-conf.tex (added), doc/jitterbuffer.txt (removed), doc/channels.txt (removed), doc/ip-tos.tex (added), doc/extensions.txt (removed), doc/queues-with-callback-members.txt (removed), doc/apps.txt (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt (removed), doc/manager.txt (removed), doc/jitterbuffer.tex (added), doc/extensions.tex (added), doc/billing.txt (removed), doc/localchannel.txt (removed), doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt (removed), doc/00README.1st (removed): Merge changes from svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc directory into a single LaTeX formatted document so that we can generate a PDF, HTML, or other formats from this information. * Add a CLI command to dump the application documentation into LaTeX format which will only be include if the configure script is run with --enable-dev-mode. * The PDF turned out to be close to 1 MB, so it is not included. However, you can simply run "make asterisk.pdf" to generate it yourself. We may include it in release tarballs or have automatically generated ones on the web site, but that has yet to be decided. 2007-03-15 18:13 +0000 [r58923] Joshua Colp * channels/chan_iax2.c: Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado) 2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant * channels/chan_sip.c: Some people like to put "limitonpeer" instead of "limitonpeers" in their configuration. While we're at it, support "limitonpeerz" and "limitonpeerssssss". (inspired by issue #9172) * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the examples section * doc/security.txt, /: Merged revisions 58896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines Add a note to the security file that the Asterisk CLI and log files may contain sensitive information, and that people should keep this in mind. ........ * configs/sla.conf.sample, apps/app_meetme.c: By default, don't attempt to do any CallerID handling at all with SLA because it is known to not work properly in some situations. However, add an option to enable it for those that would like to use it anyway. The short story behind this is that to properly handle CallerID with SLA, we need the ability to change the CallerID on an existing call, and we are not ready to handle that. 2007-03-14 01:47 +0000 [r58880] Tilghman Lesher * funcs/func_strings.c: Issue 9162 - pbx_substitute_variables_helper assumes the buffer is initialized to all zeroes. This fixes a case where it wasn't. 2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant * apps/app_meetme.c: Ensure that the blinky lights show that the trunk stopped ringing when the trunk hangs up before a station has answered it. (issue #9234, reported by francesco_r) * configs/sla.conf.sample: fix the reference to the SLA documentation 2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan) ........ * channels/chan_sip.c: Don't hangup the call on OK or errors on MESSAGE and INFO inside of a dialog (like video update requests). * channels/chan_sip.c: Issue #9251 - Clear From URI from user attributes (tgrman) 2007-03-12 16:52 +0000 [r58833] Joshua Colp * /: Blocked revisions 58832 via svnmerge ........ r58832 | file | 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't use the assembler version of fetchadd_int under Intel Macs. (issue #9254 reported by darrell budic) ........ 2007-03-12 13:08 +0000 [r58825-58826] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 57034,57523,57753,58558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line fixed another place where the out_cause was hardcoded to 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line we can free channel 31 as well, since we can occupy it ........ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: added UU transceiving and corect handling for rdnis 2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp * main/rtp.c: Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ) * channels/chan_sip.c, configs/sip.conf.sample: Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska) 2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant * channels/chan_iax2.c: Fix a few more places in chan_iax2 where the ast_frame used for receiving a frame was not properly initialized. - Interpolating a frame when the jitterbuffer is in use - decrypting a frame when IAX2 encryption is on - frames in an IAX2 trunk * apps/app_meetme.c: Make the compiler happy and initialize a variable. * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added): Merge some updates to the SLA documentation. I plan to keep working on this to explain all of the expected behavior with call handling, configuration details for specific phones, and other things. However, I got tired of doing it in plain text, so I switched to using LaTeX. I have included the PDF version. I haven't been able to get a nice looking plain text version out of it yet, but I'm not terribly concerned since this is supposed to be more of the manual, while the plain text sample configuration file is the reference. 2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp * apps/app_voicemail.c: Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai) * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done. ........ 2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant * apps/app_meetme.c: Hang up the channel that put the call on hold in the event processing thread to avoid a race condition. Also, if the station originated the call that it is putting on hold, don't hang up the trunk if it was the only station on the call and it is hanging up due to hold and not a normal hangup. * channels/chan_zap.c: Add a missing break statement so that handling the above event does not incorrectly destroy the channel. (issue #9242, andrew) 2007-03-08 21:33 +0000 [r58479] Tilghman Lesher * res/res_odbc.c: Fix segfault (Issue 9236) 2007-03-08 20:54 +0000 [r58474] Russell Bryant * apps/app_meetme.c: Refactor hold handling a bit so that it does not require keeping the call up when a call is put on hold. 2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp * main/rtp.c: Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu) * main/dsp.c, /: Merged revisions 58388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez) ........ 2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming * main/http.c: this change was not needed; fclose() handles closing the file descriptor already * apps/app_meetme.c: fix a compiler warning, and overwriting 'res' value * main/http.c: fix two cases where HTTP session file descriptors would not be closed 2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant * channels/chan_zap.c, configure, configure.ac: If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED. * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey Moore) Merged revisions 58242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines Fix a problem where the Asterisk channel name could be that of the wrong IAX2 user for a call. This is because the first step of choosing this name is to look for an IAX2 peer that happens to have the same IP/port number that this call is coming from and assuming that is it. However, this is not always correct. So, I have made it change this name after authentication happens since at that point, we have an exact match. ........ 2007-03-07 17:52 +0000 [r58240] Joshua Colp * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) 2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant * /: Blocked revisions 58167 via svnmerge ........ r58167 | russell | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a misplaced block of code in the 1.2 version of the patch to fix issue #8977 ........ * main/manager.c, /: Merged revisions 58164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines If the channels acquired using the manager Redirect action are not up, then don't attempt to do anything with them. It could lead to weird behavior, including crashes. (issue #8977) ........ 2007-03-06 23:10 +0000 [r58121] Steve Murphy * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null. ........ 2007-03-06 23:00 +0000 [r58119] Russell Bryant * configs/voicemail.conf.sample: Clarify the documentation of the dialout and sendvoicemail options. (issue #9000, caio1982 and serge-v) 2007-03-06 20:37 +0000 [r58053] Olle Johansson * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines Change error message to proper message ........ 2007-03-06 18:01 +0000 [r58023] Russell Bryant * channels/chan_skinny.c: Return an error of transmit_response is called without a session. (issue #9002) 2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp * channels/chan_iax2.c: Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh) * apps/app_meetme.c: Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r) * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares) ........ 2007-03-05 15:20 +0000 [r57826] Steve Murphy * main/pbx.c, /: Merged revisions 57825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong) ........ 2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp * main/slinfactory.c: Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj) * res/res_jabber.c: Don't reference a potentially NULL pointer. (issue #9199 reported by klolik) * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg) 2007-03-03 15:31 +0000 [r57707] Steve Murphy * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7: Updated the regression tests 2007-03-03 06:45 +0000 [r57649] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines Memory leak of a list, if call recording was abandoned ........ 2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard * main/say.c: submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri 2007-03-03 00:02 +0000 [r57591] Russell Bryant * configs/sla.conf.sample: add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\! 2007-03-02 Russell Bryant * Asterisk 1.4.1 released. 2007-03-02 23:03 +0000 [r57556] Russell Bryant * configure, configure.ac: Update the check that is used to determine whether zaptel transcoder support is present. The interface has changed. 2007-03-02 17:06 +0000 [r57477] Joshua Colp * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error. ........ 2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy * main/pbx.c, /: Merged revisions 57458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction ........ * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes right, but 9184 points out the problem-- the escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL. 2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant * main/file.c: Return the correct digit that interrupted the stream. This fixes exiting the Background application when using the m option. (issue #9176, mjagdis) * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt, include/asterisk/channel.h: Merge changes from svn/asterisk/team/russell/sla_updates * Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. 2007-03-01 22:21 +0000 [r57318] Joshua Colp * channels/chan_local.c, /: Merged revisions 57317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique) ........ 2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. 2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla docs * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes from svn/asterisk/team/russell/sla_updates * Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. * apps/app_meetme.c: Minor formatting change * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from svn/asterisk/team/russell/sla_updates * Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. 2007-02-28 19:23 +0000 [r57139] Steve Murphy * main/pbx.c, /: Merged revisions 57118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails ........ 2007-02-28 18:57 +0000 [r57093] Joshua Colp * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes) ........ 2007-02-28 18:20 +0000 [r57089] Russell Bryant * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of changes from svn/asterisk/team/russell/sla_updates * Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks|stations)" CLI commands. 2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp * apps/app_meetme.c: Picky compiler... * apps/app_speech_utils.c: Better handle timeouts when the individual speaks after everything has been played but before the timeout ends. 2007-02-28 17:15 +0000 [r57049] Steve Murphy * pbx/pbx_ael.c: I was surprised that I had not yet downgraded missing goto targets and macro call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code. 2007-02-27 17:36 +0000 [r56975] Russell Bryant * apps/app_voicemail.c: Fix voicemail email attachments. I missed the conversion of one of the line endings and there was an extra one where it should not have been. (issue #9128) 2007-02-26 22:01 +0000 [r56922] Tilghman Lesher * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky, picky... show deprecation warning in application help, too (reported via list) 2007-02-26 20:42 +0000 [r56888] Russell Bryant * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where if a device was not specified in alsa.conf, then we just use the system default, instead of creating our own default of hw:0,0. (issue #9139) 2007-02-26 20:07 +0000 [r56856] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 lines Obey the clearglobalvars option in extensions reload (or dialplan reload depending on your version). (issue #9146 reported by ramonpeek) ........ 2007-02-26 20:04 +0000 [r56847] Russell Bryant * channels/chan_iax2.c: Fix a crash in my last change to iax2_indicate(). (issue #9150) 2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp * apps/app_record.c: Update app_record documentation to use new CLI command, core show file formats. (issue #9151 reported by junky) * main/pbx.c: Use ast_strlen_zero to see if the language and/or context argument is not present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis) 2007-02-26 16:51 +0000 [r56785] Russell Bryant * channels/chan_iax2.c: Do more complete locking of the chan_iax2_pvt struct in the indicate callback. (Problem brought up by Ben Smithurst on the asterisk-dev list) 2007-02-26 16:36 +0000 [r56783] Joshua Colp * main/asterisk.c: Allow both of the show version files and core show file versions CLI commands to work. (issue #9135 reported by mvanbaak) 2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant * apps/app_meetme.c: Move a comment to be in the correct struct. * /: Blocked revisions 56729 via svnmerge ........ r56729 | russell | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure that lock.h is included in utils.c with AST_API_MODULE defined so that the implementations will be properly included when the AST_INLINE_API functions are not going to be inlined. (issue #9124, festr) ........ 2007-02-25 14:46 +0000 [r56685] Tilghman Lesher * main/channel.c, /: Merged revisions 56684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the channel list, then evaluating additional conditions (e.g. name prefix) will cause a NULL dereference. ........ 2007-02-24 02:02 +0000 [r56569] Jason Parker * channels/chan_skinny.c: Make sure to set a speeddials parent on creation. Don't crash if hold is pressed when no call is active. Don't return in places that we shouldn't.. 2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming * codecs/codec_zap.c: update to match zaptel 1.4 API change that was committed a few minutes ago 2007-02-23 23:24 +0000 [r56505] Russell Bryant * main/asterisk.c, /: Merged revisions 56504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines Fix up a couple more signal handlers to not do bad things that could cause various undesirable results. The other day, I made Asterisk deadlock by hitting Control-C because of a bad signal handler. Now, signal handlers just set a flag and write to an alert pipe for the flag to be handled. Then, there is another thread that is monitoring for these flags. If being run in console mode, it is just the main thread. If Asterisk is in the background, a thread is created to do it. ........ 2007-02-23 21:53 +0000 [r56457] Joshua Colp * main/sched.c: Change log notice to debug. It is possible for a scheduled item to execute and be deleted at close to the same time and unavoidable. If this happens this message creeps up. 2007-02-23 20:20 +0000 [r56407] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines Don't destroy mutexes before unregistering all of the entry points from the core. Also, fix a potential memory leak from not destroying the locks for all of the possible call numbers (about 32k of them). ........ 2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming * build_tools/make_version_h: build special version strings for AADK/S800i builds 2007-02-23 17:58 +0000 [r56341] Russell Bryant * apps/app_voicemail.c: The IMAP storage code uses the same code to build the email that is used when voicemail is sent via email using something like sendmail. In the patch from bug 8033 to fix various IMAP storage problems, the line endings in the email file were changed in the code from "\n" to "\r\n". However, this breaks sending regular voicemail to email. So, this change conditionally sets line endings to "\r\n" only if IMAP_STORAGE is enabled. (issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage) 2007-02-22 23:25 +0000 [r56280] Joshua Colp * /: Blocked revisions 56279 via svnmerge ........ r56279 | file | 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always defer Agent logoff if any channels are up until they hang up. (issue #9123 reported by arbrandes) ........ 2007-02-22 23:08 +0000 [r56277] Russell Bryant * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c, doc/sla.txt: Merge changes from team/russell/sla_updates. This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. 2007-02-22 18:49 +0000 [r56231] Joshua Colp * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel. ........ 2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development * doc/osp.txt: Update OSP documentation for v1.4. 2007-02-22 10:33 +0000 [r56125] Olle Johansson * channels/chan_sip.c: Move message from verbose to debug 2007-02-22 02:39 +0000 [r56094] Steve Murphy * sounds/Makefile: updated the sound tarball versions in Makefile 2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant * channels/chan_sip.c: Restructure a little bit of code to reduce nesting. There is no functionality change here. * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines If we receive a frame that is not in any of the negotiated formats, then drop it. (potentially issue #8781 and SPD-12) ........ 2007-02-22 00:35 +0000 [r56008] Joshua Colp * main/cli.c: Print out deprecation notice on usage output of CLI commands. (issue #8925 reported by blitzrage) 2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming * main/loader.c: disable unloading of embedded modules... there is a fundamental problem with doing so that will not be fixed in this version of Asterisk due to its invasiveness 2007-02-21 20:35 +0000 [r55957] Joshua Colp * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message. ........ 2007-02-21 20:27 +0000 [r55954] Jason Parker * channels/chan_gtalk.c: Fix locking issue, and accept "transport-accept" as a valid accept message. This should solve issues 8970 and 8503. 2007-02-21 20:22 +0000 [r55951] Russell Bryant * apps/app_meetme.c: Simplify the last change to app_meetme, and move the call to dispose_conf() up into the block where we know a conf exists. 2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp * apps/app_meetme.c: Only dispose of the conference if one was created. * apps/app_speech_utils.c: Only start playing the next file if we have not been quieted. * channels/chan_sip.c: Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113) 2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming * /, build_tools/make_version: Merged revisions 55868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 Feb 2007) | 2 lines use new tag version script ........ 2007-02-21 08:32 +0000 [r55834] Olle Johansson * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg) 2007-02-21 02:01 +0000 [r55799] Jason Parker * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found. Issue 7764, patch by sailer 2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant * apps/app_meetme.c: Improve the reference counting to fix bugs where people report seeing conferences listed that have no members. (issue #9073) * /: Blocked revisions 55750 via svnmerge ........ r55750 | russell | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix random crashes when using the MeetMe application. This patch converts list handling to use the linked list macros and most importantly, implements reference counting on the ast_conference objects. The reference counting was first backported from 1.4. However, that code has some problems that caused the reference count to never hit zero. Those problems are fixed in this patch and will be resolved in 1.4 and trunk next, with a different patch. (issues #7647, #9073, #9106, BE-115). ........ 2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp * apps/app_voicemail.c: Better handle dropped IMAP connections. (issue #9054 reported by bsmithurst) * channels/chan_sip.c: Return behavior I removed. I did not remember that you could just add a localnet entry to make it work. * channels/chan_sip.c: Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska) * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155) ........ 2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant * main/http.c: Add the Asterisk version information to the Server header in HTTP responses. (requested by Pari) * include/asterisk/manager.h: Increase the maximum number of manager headers to 128, at the request of Pari. * /: Blocked revisions 55588 via svnmerge ........ r55588 | russell | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert a tab to spaces so that the documentation is printed out properly aligned. ........ 2007-02-20 16:53 +0000 [r55555] Jason Parker * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free with strdupa (thanks file) 55555! 2007-02-20 16:41 +0000 [r55553] Russell Bryant * configs/sla.conf.sample: Change the formatting of sla.conf.sample to make it more readable. (issue #9112, blitzrage) 2007-02-19 21:12 +0000 [r55483] Olle Johansson * res/res_jabber.c: - Not sending arguments to an application is not "out of memory" - Making error messages a bit more clear 2007-02-19 18:11 +0000 [r55435] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) | 2 lines forcename and forcegreetings options should check to see if the recording already exists ........ 2007-02-19 14:52 +0000 [r55397] Doug Bailey * channels/chan_iax2.c: Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit. Modified module unload process to avoid deadlocks on pthread cancels 2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson * /, apps/app_record.c: Merged revisions 55277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 lines Documentation update (#9053, jsmith) ........ * /: Block patch that was made only for 1.2 (already implemented in 1.4 and trunk) 2007-02-17 17:39 +0000 [r55219] Joshua Colp * apps/app_queue.c: Add missing membername option to AddQueueMember documentation. (issue #9088 reported by seanbright) 2007-02-17 17:10 +0000 [r55217] Jason Parker * channels/chan_skinny.c: Fix an issue where callerid would not be displayed on some phones. Issue 8995, initial patch and research done by wedhorn 2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines Answer the channel before recording privacy information. (issue #8926 reported by lmamane) ........ * apps/app_queue.c: Make the 'i' option of Queue actually work. (issue #8986 reported by utis) * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba) ........ 2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant * configure, include/asterisk/autoconfig.h.in, configure.ac: If the pg_config application is found, but there is probably executing it, then consider postgres unavailable. (issue #8637) * codecs/gsm/Makefile: Filter out yet another architecture that does not work with the optimizations in the built-in libgsm. (issue 8637, ovi) * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged revisions 55005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a conference is created from meetme.conf, it is acceptable behavior that the pin can not be changed until the conference goes away. I also added a note in meetme.conf to describe this behavior. We still have another issue in 1.4 and trunk where some conferences with no users don't go away. That is the real bug that needs to be addressed here. ........ 2007-02-16 22:18 +0000 [r55002] Joshua Colp * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode) ........ 2007-02-16 21:12 +0000 [r54969] Russell Bryant * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines For conferences that are configured in meetme.conf, check the configuration file every time someone joins the conference instead of only when the conference is first created. This is to ensure that changes to the pin numbers in the config file are always honored. (issue #9073) ........ 2007-02-16 18:51 +0000 [r54924] Joshua Colp * apps/app_dial.c: Need to check macro extension as well as macro context for directed pickup. 2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by default. It was set to enabled in pbx.c. However, if the option was not present in extensions.conf, then pbx_config.c would set it back to disabled. * res/res_features.c: Clean up a few coding guidelines issues - spaces to tabs, use sizeof() to pass the size of a static buffer, add spaces ... 2007-02-16 17:25 +0000 [r54886] Jason Parker * main/asterisk.c: Clarify a restart message. It's silly, but the reporter had a very valid point. Issue 9079 2007-02-16 17:02 +0000 [r54884] Joshua Colp * apps/app_dial.c: Allow directed pickup to pick up the real context instead of the macro context if a Macro is used. (issue #8984 reported by jamesb63) 2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson * channels/chan_sip.c: Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted. * /, res/res_agi.c: Merged revisions 54771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 lines Issue #9069 - If we open with TH we should not close with /TD. (seanbright) ........ 2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp * apps/app_speech_utils.c: Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek) * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines Use a separate variable to indicate execution should continue instead of the return value. (issue #8842 reported by pluto70) ........ * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman) 2007-02-14 18:44 +0000 [r54439] Olle Johansson * /: Block patch only needed in 1.2 2007-02-14 16:56 +0000 [r54375] Matt Frederickson * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011. ........ 2007-02-14 01:09 +0000 [r54290] Joshua Colp * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork) 2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant * channels/chan_sip.c: Remove a couple of leftover debug messages * include/asterisk/devicestate.h: Fix the documentation on the return values from device state provider registration and deletion. * channels/chan_sip.c: If we fail to create the SIP socket, then return -1 from reload_config() so that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get spammed with error messages every time chan_sip tries to send a message. 2007-02-13 18:41 +0000 [r54180] Olle Johansson * /: Blocking patch for 1.2 only 2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant * main/dial.c, include/asterisk/dial.h: Change ast_set_state_callback() to ast_dial_set_state_callback() * main/dial.c, apps/app_meetme.c, apps/app_page.c, include/asterisk/dial.h: - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API 2007-02-12 16:34 +0000 [r54026] Joshua Colp * configure, configure.ac: Make the --without-oss argument work. (issue #9026 reported by puzzled) 2007-02-12 15:38 +0000 [r54002] Russell Bryant * configs/users.conf.sample: Fix a typo where "vmpassword" should be "vmsecret" 2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach * channels/chan_h323.c: Fix VLDTMF reception * apps/app_echo.c: Much simpler than previous one ;-) * main/channel.c: Provide correct DTMF duration * main/cli.c: Bring deprecated 'debug channel ' command back 2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming * configure, configure.ac, acinclude.m4: don't display the --with-imap message unless --with-imap was specified without a path use '-n' instead of '! -z' for tests 2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant * apps/app_meetme.c: Add some output for "show application SLAStation/SLATrunk" * channels/chan_sip.c: Change some text to properly state "On Hold", which was already done in trunk. * configs/sla.conf.sample, include/asterisk/app.h, include/asterisk/utils.h, main/dial.c, apps/app_meetme.c, channels/chan_sip.c, doc/sla.txt (added), include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge team/russell/sla_rewrite This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. * apps/app_echo.c: When the Echo() application receives the digit '#', echo that back as well. Since we already sent the BEGIN frame for that digit, it makes sense to send the END as well. 2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming * channels/chan_gtalk.c: another dependency * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c, funcs/func_odbc.c, res/res_adsi.c: add some inter-module dependencies * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file 2007-02-09 19:33 +0000 [r53749] Joshua Colp * apps/app_dial.c: Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee) 2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming * doc/imapstorage.txt, configure, configure.ac: clarify the fact that voicemail IMAP storage cannot be built against a distro's binary c-client library package (at least not at this time) 2007-02-08 23:18 +0000 [r53672] Olle Johansson * main/acl.c: Don't output debug unless we asked for it 2007-02-08 17:54 +0000 [r53601] Joshua Colp * apps/app_speech_utils.c: Fix timeout issue when utterance is longer then timeout itself. 2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher * main/loader.c: Issue 9007 - Mutex not released on early return * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) | 2 lines Issue 9003 - If fullname is empty, quote() passes back "\"" ........ 2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant * main/db1-ast/Makefile: When building libdb1.a, put the additional flags needed at the beginning of ASTCFLAGS, instead of at the end. This way, we ensure that we find the local headers first before accidentally trying to use headers that exist in locations specified in the ASTCFLAGS passed from the main Makefile. (issue #8637, ovi) * main/Makefile: The clean target actually needs to run "distclean" on editline. This is because we need to make sure that its configure script gets executed again, because the CFLAGS we want to pass to editline may have changed. 2007-02-07 17:53 +0000 [r53434] Joshua Colp * main/rtp.c: We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982) 2007-02-07 17:39 +0000 [r53429] Russell Bryant * main/rtp.c: When parsing the NTP timestamp in a sender report message, you are supposed to take the low 16 bits of the integer part, and the high 16 bits of the fractional part. However, the code here was erroneously taking the low 16 bits of the fractional part. It then shifted the result 16 bits down, so the result was always zero. This fix makes it grab the appropriate high 16 bits, instead. (issue #8991, pointed out by andre_abrantes) 2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp * apps/app_playback.c: Directly load say.conf in load_module instead of calling the reload function. (issue #8946 reported by junky) * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst) ........ 2007-02-07 15:33 +0000 [r53355] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) | 2 lines Issue 7440 - Macro called from Macro from the h extension exits prematurely ........ 2007-02-07 09:22 +0000 [r53324] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 52843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated) ........ 2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher * res/res_jabber.c: Text fix for jabber reload command (reported by bkruse via IRC) * main/manager.c, /: Merged revisions 53245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) | 2 lines Issue 8987 - Status could return two responses (mnicholson) ........ 2007-02-05 23:43 +0000 [r53222] Olle Johansson * channels/chan_sip.c: Formatting 2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp * apps/app_playback.c: Ensure say_cfg is NULL when the module is loaded. (issue #8946 reported by junky) * apps/app_playback.c: Unregister Playback CLI commands as well as dialplan application. (issue #8946 reported by junky) 2007-02-05 00:18 +0000 [r53143] Olle Johansson * channels/chan_sip.c: Add some comments on queue system behaviour and how it affects the SIP channel 2007-02-03 21:05 +0000 [r53138] Joshua Colp * channels/chan_sip.c: Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113) 2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application exits early because of invalid arguments instead of just leaving it empty. (issue #8975) ........ * /: Blocked revisions 53134 via svnmerge ........ r53134 | russell | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert some changes that accidentally got committed as a part of another fix. ........ 2007-02-03 10:02 +0000 [r53131] Paul Cadach * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string because due to compatibilities with CS1000 reported at www.voip-info.org 2007-02-02 21:26 +0000 [r53129] BJ Weschke * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a warning to the console that things might possibly be misconfigured when queue member's states are still 'Not in Use' when we're about to bridge them with a caller from queue. Also, put some documentation quoted from oej's queues.txt efforts started in /trunk today. This commit puts #7433 into feedback state for 1.4, and pending no further negative feedback, it will finally be closed. 2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp * main/rtp.c: Correct a copy/pasted error message line for RTCP. * main/config.c, /: Merged revisions 53117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 lines Pass the glob expanded filename to process_text_line so that error messages contain the actual filename, not the original include one. (issue #8959 reported by tzafrir) ........ * Makefile: Add systemname to asterisk.conf generation per recent discussions about it. (issue #8968 reported by blitzrage) 2007-02-02 00:24 +0000 [r53109] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. 2007-02-01 23:16 +0000 [r53108] Jason Parker * /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell | 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a small typo. Synopsis lines shouldn't have a newline ........ 2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE. ........ * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile here as well, but it apparently required both dev mode and no optimizations to creep up. * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) ........ 2007-02-01 21:24 +0000 [r53093] Russell Bryant * funcs/func_strings.c: Fix the FIELDQTY function to not crash. (reported by blitzrage and Corydon on IRC) 2007-02-01 21:15 +0000 [r53091] Olle Johansson * /: Going backwards, blame file. 2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2 lines Return previous behavior of having MOH pick up where it was left off. (issue #8672 reported by sinistermidget) ........ * funcs/func_strings.c: Make func_strings build under dev mode. Didn't I do this today already in the berkeley DB? 2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement call counter - If it's still set at time of dialog destruction, make sure we decrement the device call counter properly before we destroy the dialog * apps/app_queue.c: Change debug level for state change message that is not really informative when debugging app_queue * channels/chan_sip.c: Cleaning up the devicestate callback function 2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher * funcs/func_strings.c: Oops. * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) | 2 lines Bug 8965 ........ 2007-02-01 19:33 +0000 [r53072] Joshua Colp * main/asterisk.c: Add missing 'F' letter to getopt so it magically becomes a valid option. (issue #8960 reported by tzafrir) 2007-02-01 19:21 +0000 [r53070] Tilghman Lesher * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) | 2 lines No wonder FIELDQTY doesn't work with functions... the documentation in pbx.c was wrong ........ 2007-02-01 17:37 +0000 [r53064] Joshua Colp * channels/chan_sip.c: Fix silly logic. We really want to write UDPTL frames out when the call is up. 2007-02-01 16:35 +0000 [r53062] Olle Johansson * configs/sip.conf.sample: Add explanation of port= in combination with defaultip= (thanks jsmith) 2007-02-01 13:17 +0000 [r53060] Christian Richter * channels/chan_misdn.c: we update the name on any first reply of our setup 2007-02-01 11:07 +0000 [r53057] Paul Cadach * channels/chan_h323.c: chan_h323 is very stable, so let it built by default 2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp * main/rtp.c: When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party. * main/rtp.c: Add more frame types to forward in the RTP bridge loops. 2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant * main/cdr.c, main/manager.c, pbx/pbx_spool.c, channels/chan_skinny.c, channels/chan_h323.c, main/http.c, pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c, main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: Merged revisions 53045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines Fix a bunch of places where pthread_attr_init() was called, but pthread_attr_destroy() was not. ........ * apps/app_userevent.c: Remove an extra \r\n from manager user events. (issue #8955, mnicholson) * main/rtp.c, /: Merged revisions 53039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines Use the proper format string to print unsigned values in the rtp debug output. (issue #8954, wmis) ........ * apps/app_queue.c: Only changed the paused status in an existing queue member if the paused column exists. * apps/app_queue.c: Instead of always creating a realtime queue member as unpaused, read the "paused" column and use that value for the paused status of the member. (issue #8949, jmls) * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10. (issue #8363, johnlange) * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue #8942, lters) * configure, include/asterisk/autoconfig.h.in, configure.ac, codecs/codec_gsm.c: When we are checking for a system installed version of libgsm, we need to check for gsm.h as well. Furthermore, when checking for this header, it may be located in a gsm/ sub directory, so check for that, as well. (issue #8773) * /: Blocked revisions 52954 via svnmerge ........ r52954 | russell | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't print a message indicating that we don't know what to do with a proceeding control frame in ast_request_and_dial(). We just need to ignore it. (reported by JerJer on #asterisk-dev) ........ * channels/chan_sip.c: Only set the DTMF flag on the rtp structure if the DTMF mode is actually RFC2833, not just that it is not INFO. This makes it get set for inband DTMF as well, which is not valid. (issue #8936) * main/asterisk.c, /: Merged revisions 52903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | 9 lines The SIGHUP handler was implemented to allow admins to send SIGHUP to a running Asterisk process to reload the configuration. However, doing the actual reload in the signal handler itself is a very bad thing to do, because the reload process includes calling non-reentrant functions such as malloc/calloc/etc. If Asterisk is running in the background, then the reload will happen immediately. However, if running in console mode, the reload doesn't work until something is typed at the console. That sort of defeats the purpose, but I don't see an easy way to get around it at this point. ........ * /: Blocked revisions 52857 via svnmerge ........ r52857 | russell | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment out the parts in the Makefile that make codec_zap get built. It will not yet build against zaptel 1.2, so I am disabling it to prevent further bug reports until it gets merged. (issue #8940) ........ 2007-01-30 15:29 +0000 [r52856] Joshua Colp * channels/chan_iax2.c: Drop the deprecated show commands since the original ones were changed back. (issue #8937 reported by PCadach) 2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach * channels/chan_h323.c: Revert reprecation of h.323 gk cycle command from pre-1.4 version instead of duplicated h323 cycle gk * res/res_odbc.c: Don't play with free()'d pointers * configure, acinclude.m4: Handle non-standard OpenH323/PWLib library names 2007-01-30 00:15 +0000 [r52763] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines Fix the extraction of the timestamp from video frames. It was using the mapping for a mini-frame instead of a video-frame, which caused it to get invalid data. (issue #8795, mihai) ........ 2007-01-29 23:43 +0000 [r52717] Joshua Colp * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev) ........ 2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant * main/Makefile: Add a missing quotation mark. This was pointed out by jcmoore on #asterisk-dev. * main/manager.c: Remove a recursive lock of the manager session. This was pointed out by zandbelt in issue #8711. 2007-01-29 22:12 +0000 [r52679] Tilghman Lesher * pbx/pbx_config.c: Argument number correction 2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant * main/Makefile: ASTLDFLAGS needs to be passed to the editline configure script as LDFLAGS. (issue #8928, zandbelt) * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P bridging can only be used when the DTMF modes don't match if the core is monitoring DTMF in both directions. Then, the core will handle the translation. Otherwise, this bridging method can not be used. (issue #8936) * main/manager.c: The session lock can not be held while calling action callbacks. If so, then when the WaitEvent callback gets called, then no event can happen because the session can't be locked by another thread. Also, the session needs to be locked in the HTTP callback when it reads out the output string. This fixes the deadlock reported in both 8711 and 8934. Regarding issue 8711, there still may be an issue. If there is a second action requested before the processing of the first action is finished, there could still be some corruption of the output string buffer used to build the result. (issue #8711, #8934) 2007-01-29 18:59 +0000 [r52572] Joshua Colp * apps/app_voicemail.c: Use ast_calloc instead of malloc. 2007-01-29 17:57 +0000 [r52535] Steve Murphy * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR backport to 1.4). It was committed to trunk via 7663. But it wasn't so much an enhancement as a fix for the bad language output for portuguese in Brazil, so, after a lot of prodding from patient Brazilians, here is the same fix for 1.4 2007-01-29 17:33 +0000 [r52523] Joshua Colp * apps/app_voicemail.c: Set quota information to 0 when creating a vm_state. (issue #8924 reported by neutrino88) 2007-01-29 16:54 +0000 [r52506] Russell Bryant * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in the last commit to the adaptive jitterbuffer code. - Specifically indicate to the compiler that the "dropem" variable only needs one but. - Change formatting to conform to coding guidelines. 2007-01-29 04:18 +0000 [r52494] Jim Dixon * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with jitterbuf, whereas it would not complain about, and would allow itself to be overfilled (per the max_jitterbuf parameter). Now it rejects any data over and above that size, and complains about it. 2007-01-28 05:15 +0000 [r52462] Tilghman Lesher * configure, configure.ac: Suggested change to fix normal usage of --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing list) 2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log follow documentation. (issue #7677 reported by amilcar) ........ * main/manager.c: Have the manager interface send back an "Already logged in" message instead of "Invalid/Unknown Command" when the client authenticates for a second time. (issue #8509 reported by pari) * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines Make the last context entry read in the dominant one. (issue #8918 reported by pj) ........ * main/file.c: Fix core show file formats CLI command. 2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 lines Allow dequeueing of frames with negative timestamp by moving jitterbuffer frames check to jb_next. (issue #8546 reported by harmen) ........ * channels/chan_sip.c: Drop out variables I accidentally put in. * channels/chan_sip.c: Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42) * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy) ........ 2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant * apps/app_voicemail.c, configs/users.conf.sample: By suggestion from kpfleming last week, change "vmpassword" to "vmsecret". * configure, configure.ac: Remove libnsl as a required lib for libiksemel to work. This change was already made in the trunk. (issue #8762) * /: Blocked revisions 52137 via svnmerge ........ r52137 | russell | 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a seg fault when running this application with no arguments from AGI. (issue #8905, junky) ........ * include/asterisk/dial.h: Fix the formatting of doxygen comments to properly indicate that the comment documents the previous entity, as opposed to the next one. 2007-01-24 18:26 +0000 [r52052] Steve Murphy * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 line updated check_expr via 8322 (refactoring of expression checking impl); elfring contributed a nice code reorg, I contributed some time to get it working again, better messages ........ 2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp * main/dial.c (added), apps/app_page.c, main/Makefile, include/asterisk/dial.h (added): Merge in dialing API and the app_page that uses it. (issue #BE-118) * channels/chan_sip.c: Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc) 2007-01-24 17:14 +0000 [r52000] Russell Bryant * configure: rebuild configure script to reflect last chan_h323 related changes. 2007-01-24 12:57 +0000 [r51979-51989] Christian Richter * channels/chan_misdn.c: added fix from #8899 * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 Jan 2007) | 1 line fixed the busy problem (dialstatus was not busy when we called a busy extension) ........ 2007-01-24 09:30 +0000 [r51931] Olle Johansson * channels/chan_sip.c: Show capabilities *and* preference in general settings in "sip show settings" (reported by Clona/Telio - Thanks!) 2007-01-24 08:04 +0000 [r51895] Paul Cadach * acinclude.m4: Allow x64 builds of H.323 (please, rebuild configure) 2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant * main/channel.c, /: Merged revisions 51843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines Fix an issue related to synchronization of recordings when using Monitor(). The bug is a miscalculation of the amount to seek the stream for writing to disk when the number of samples coming in and out of a channel do not match up. (issue #8298, #8887, report and patch by guillecabeza, patch files created and testing done by whoiswes) ........ * apps/app_while.c, /: Merged revisions 51828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | 4 lines Don't set a new value for the END_ variable on the channel before using the old value. If you do, it will lead to accessing a memory address that has been free()'d. (issue #8895, arkadia) ........ 2007-01-23 22:46 +0000 [r51788] Joshua Colp * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_alsa.c, channels/chan_gtalk.c, channels/chan_iax2.c: Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky) 2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant * main/manager.c: Fix some bugs in process_message(). The manager session lock needs to be held when sending some sort of response, or calling one of the manager action callbacks. This resolves an issue where people using the GUI would get random crashes when they start clicking around a lot. (issue #8711, reported and debugged by zandbelt) * main/http.c: Fix setting the default port of 8088 on 64-bit or big-endian machines. * main/manager.c: When traversing the list of manager actions, the iterator needs to be initialized to the list head *after* locking the list. Also, lock the actions list in one place it is being accessed where it was not being done. 2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy * res/res_features.c: this mod from 8593 (dstchannel in cdr is empty when transfer call). * main/callerid.c: via 8748 (callerid.c loses name when returning PRIVATE_NUMBER flag), the user suggested this mod, saying it would allow 'WITHHELD' to appear in the name field, which would be useful 2007-01-23 10:28 +0000 [r51648-51649] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | 6 lines * more additions to make the RESTART message work * added fix for misdn_call to allow SETUPs with empty extensions, replaced the strtok_r functions with strsep for that (inspired by Sandro Cappellazzo, thanks) ........ r50506 | crichter | 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get L2 UP, the L1 is UP definitely too, so we set the L1 state up as well. ........ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c: manually merged r49922 and r50335, because of conflicts. this commint includes addition of the ISDN RESTART Message 2007-01-23 06:51 +0000 [r51615] Paul Cadach * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen) 2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp * channels/chan_sip.c: Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc) * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan 2007) | 2 lines Yield before reading from zaptel timing source under Solaris so that other threads get a chance to do things. (issue #7875 reported by bob) ........ 2007-01-22 19:41 +0000 [r51411] Russell Bryant * /: Blocked revisions 51410 via svnmerge ........ r51410 | russell | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge codec_zap support for the transcoder card. This is a standalone codec module so it will not affect anything else. ........ 2007-01-22 19:28 +0000 [r51409] Steve Murphy * pbx/pbx_ael.c: This fixes 8836, according to dnatural 2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil) ........ * /: Blocked revisions 51359 via svnmerge ........ r51359 | file | 2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly declare what codecs are supported by default globally since using a bitmask for all may include ones we don't need. (issue #8357 reported by gknispel_proformatique) ........ 2007-01-20 06:53 +0000 [r51348-51350] Jason Parker * configs/say.conf.sample: Fix Italian numeral support in say.conf for "_[2-9]00" case. "2131" would've translated to something along the lines of (pardon my..Italian {or lack thereof}) "duecentocentotrentuno", which makes no sense at all. * configs/say.conf.sample: Fix German language support in say.conf Properly support 21, 31, 41, 51, 61, 71, 81, and 91. einundzwanzig has the same format as zweiundzwanzig (as do all other "_ZX" spoken numerals) Fix support for numbers in the 10,000,000 to 99,999,999 range. Add support for numbers in the 100,000,000 to 999,999,999 range. 2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant * apps/app_meetme.c: Remove an unused instance of an unnamed enum. * apps/app_meetme.c: Remove another duplicated definition * apps/app_meetme.c: Remove a variable that was declared twice. * codecs/gsm/Makefile: Add a couple more processors that need optimizations excluded. (issue #8637) * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same thing. So, a digit would have been interpreted incorrectly here. Since the channel driver will always have the begin and end callbacks called for a digit, only support the button-down and button-up messages. * .cleancount: Bump the cleancount since my last commit changed the channel structure. * channels/chan_oss.c, main/rtp.c, main/channel.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_local.c, main/frame.c, channels/chan_sip.c, channels/chan_agent.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c: Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) * main/asterisk.c: Merged revisions 51300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | 4 lines Fix a memory leak on command line tab completion. The container for the matches was freed, but the individual matches themselves were not. (issue #8851, arkadia) ........ 2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard * channels/chan_zap.c: chan_zap compiles without libpri after committing 7877 patch * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file. ........ 2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming * /: block this patch since it is already here 2007-01-18 22:50 +0000 [r51265] Jason Parker * apps/app_voicemail.c, main/channel.c, main/pbx.c, funcs/func_strings.c, main/app.c: Add some more checks for option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, patch(es) by tgrman 2007-01-18 21:54 +0000 [r51262] Russell Bryant * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in: Ensure that the locations given to the Asterisk configure script for ncurses, curses, termcap, or tinfo are further passed along to the editline configure script. This fixes some cross-compilation environments. (issue #8637, reported by ovi, patch by me) 2007-01-18 21:14 +0000 [r51256] Tilghman Lesher * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 Jan 2007) | 2 lines If a timezone is not specified, assume localtime (instead of gmtime) (Issue #7748) ........ 2007-01-18 19:17 +0000 [r51251] Joshua Colp * apps/app_speech_utils.c: Only start timeout once we reach the end of the files to play back. 2007-01-18 18:42 +0000 [r51245] Jason Parker * main/cli.c: Fix an issue with file name completion in "module load" and "load". Issue 8846 2007-01-18 18:36 +0000 [r51243] Joshua Colp * channels/chan_sip.c: Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113) 2007-01-18 18:28 +0000 [r51241] Jason Parker * main/channel.c: Fix an issue with deprecated commands 2007-01-18 17:49 +0000 [r51236] Tilghman Lesher * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 Jan 2007) | 2 lines Document all the fields, including the indication that "uniqueid" should not be renamed. ........ 2007-01-18 17:18 +0000 [r51233] Russell Bryant * main/manager.c: Make the "hasmanager" option in users.conf actually have an effect. (issue #8740, LnxPrgr3) 2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp * apps/app_voicemail.c: Build the IMAP remote directory string better and properly. Fix an issue with encoding the GSM voicemail when attaching to the voicemail. (issue #8808 reported by akohlsmith) * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by mdu113) 2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant * funcs/func_odbc.c: Fix some instances where when loading func_odbc, a double-free could occur. Also, remove an unneeded error message. If the failure condition is actually a memory allocation failure, a log message will already be generated automatically. * channels/chan_zap.c: Instead of dividing the offset by 2 directly, make it more clear that the offset is being scaled by the size of the elements in the buffer. (Inspired by a discussing on the asterisk-dev list about this code) * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines Move the check for a failure of ast_channel_alloc() to before locking the pvt structure again. Otherwise, on a failure, this will cause a deadlock. ........ 2007-01-17 20:56 +0000 [r51195] Tilghman Lesher * /, main/utils.c: Merged revisions 51194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) | 4 lines When ast_strip_quoted was called with a zero-length string, it would treat a NULL as if it were the quoting character (and would thus return the string in memory immediately following the passed-in string). ........ 2007-01-17 17:36 +0000 [r51186] Jason Parker * apps/app_voicemail.c: re-add "password" for realtime voicemail 2007-01-17 06:36 +0000 [r51182] Joshua Colp * main/rtp.c: Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna) 2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming * apps/app_voicemail.c: a few more coding style cleanups and one bug fix (from AnthonyL) 2007-01-17 00:46 +0000 [r51172] Joshua Colp * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the scheduler callback. 2007-01-17 00:20 +0000 [r51165-51170] Jason Parker * main/rtp.c: Fix issue with dtmf continuation packets when the dtmf digit is 0... Issue 8831 * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with IMAP storage and realtime voicemail. Also update the vmdb sql script for IMAP specific options. Issue 8819, initial patches by bsmithurst (slightly modified by me) * doc/voicemail_odbc_postgresql.txt: change documentation to reflect new procedure in 1.4/trunk 2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions 51161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) | 2 lines Add documentation walkthrough on getting Postgres to work with voicemail (from Issue 8513) ........ * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) | 2 lines Postgres driver doesn't like a NULL pointer when retrieving the length (Bug 8513) ........ 2007-01-16 17:46 +0000 [r51150] Matt O'Gorman * apps/app_voicemail.c: minor things i missed before i get jumped on 2007-01-16 17:39 +0000 [r51148] Joshua Colp * /, res/res_features.c: Merged revisions 51145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 lines Return previous behavior. ParkedCalls will be able to do DTMF based transfers again. trunk however will get an option to allow this to be set on/off. (issue #8804 reported by nortex) ........ 2007-01-16 17:36 +0000 [r51146] Jason Parker * main/file.c: Display more useful output when streaming files. Include the channel name to which the file is being played. Issue 8828, patch by junky. 2007-01-16 05:55 +0000 [r51087] Joshua Colp * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer) ........ 2007-01-16 01:15 +0000 [r51057] Russell Bryant * main/config.c: It is possible for the config pointer to be NULL here, so it needs to be checked before dereferencing it. 2007-01-16 00:22 +0000 [r51030] Matt O'Gorman * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for changing voicemail password in users.conf from voicemail main, written by AnthonyL bug #8436 2007-01-15 23:49 +0000 [r50994] Russell Bryant * Makefile.rules: Filter out a few CFLAGS that are not valid CXXFLAGS. 2007-01-15 23:10 +0000 [r50988] Tilghman Lesher * /: Blocked revisions 50987 via svnmerge ........ r50987 | tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines Check return value before dereferencing (Bug 8822) ........ 2007-01-15 21:08 +0000 [r50957] Matt O'Gorman * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 lines Solves issue with forwarding voicemails from folders other than inbox. patch by anthonyl. ........ 2007-01-15 18:23 +0000 [r50921] Jason Parker * main/asterisk.c: re-add deprecated "show version" CLI command. 2007-01-15 16:36 +0000 [r50895] Joshua Colp * main/manager.c: Move event processing into do_message so that it gets executed again when events are tripped. 2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming * configure, include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the ACX_PTHREAD macro from the Autoconf macro archive for setting up compiler pthreads support... should improve portability to platforms with unusual pthreads requirements 2007-01-14 21:59 +0000 [r50820] Joshua Colp * main/astmm.c: Add missing newlines for two memory CLI commands. 2007-01-14 05:13 +0000 [r50782] Tilghman Lesher * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c, main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c, main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c, main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 Jan 2007) | 2 lines Bug 8814 - db should look for its header using a relative path, instead of the system path (Fixes FreeWRT) ........ 2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming * Makefile, build_tools/make_sample_voicemail (added): when building the sample greetings for maibox 1234@default during 'make samples', build a greeting for each language and file format the user selected to install with menuselect (reported by Brian Capouch on asterisk-dev) 2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp * main/channel.c: Only write a frame out to the channel if one exists. There are cases where one may not and would therefore cause the channel driver to segfault. (issue #8434 reported by slimey) * res/res_snmp.c: Only join the snmp thread on an unload if the thread is actually running. (issue #8810 reported by junky) 2007-01-12 19:24 +0000 [r50647] Jason Parker * configs/voicemail.conf.sample: Update documentation to state that you shouldn't use realtime static with voicemail.conf 2007-01-12 16:42 +0000 [r50602] Joshua Colp * main/manager.c: We need to check for res being 0 in do_message itself, otherwise our headers will get lost. 2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming * main/pbx.c, /: Merged revisions 50561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) | 2 lines minor documentation clarification ........ 2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp * channels/chan_sip.c: Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing. * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) * apps/app_speech_utils.c: Merge speech-multi branch which adds support for joining multiple sound files together to be played one after another in SpeechBackground. * main/config.c: Fix parsing when using something like ldap settings. (done by anthonyl) * channels/chan_sip.c: Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker) 2007-01-10 16:45 +0000 [r50346] Jason Parker * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made it fail to load if the config file existed. Issue 8777 2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 lines Add another return value to dial_exec_full that indicates execution is going to continuing at a new extension/context/priority and to just let it slide. (issue #8598 reported by jon) ........ * main/pbx.c: Ensure data's existence before trying to access it. (issue #8774 reported by rcourtna) 2007-01-10 02:17 +0000 [r50228] Russell Bryant * Makefile, /: Merged revisions 50227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | 6 lines Make the number that represents the major version number a single digit instead of 2. Using two digits makes it an octal number when put into version.h, which breaks the compilation of any out of tree module that checks the version for any version after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev mailing list, who gave credit to vihai for pointing it out) ........ 2007-01-09 17:11 +0000 [r50186] Jason Parker * main/cli.c: Re-add CLI command that should have only been deprecated in 1.4. Thanks kshumard! (reported in person, so no associated issue #) 2007-01-09 13:40 +0000 [r50151] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) | 4 lines The advent of realtime has enabled people to use commas in the fullname field. This could cause an issue with sending voicemails, when the field is unquoted. (Issue 8595) ........ 2007-01-09 11:25 +0000 [r50124] Olle Johansson * channels/chan_sip.c: - handle re-invites properly in sip_hangup() - Add some invitestate status changes just to be sure 2007-01-08 23:39 +0000 [r50098] Jason Parker * apps/app_voicemail.c: Fix an issue with voicemail and users.conf, where it wouldn't ever parse a password, since it was using "secret" instead of "password" Issue 8761, reported by and patch suggestion from ssokol. 2007-01-08 21:11 +0000 [r50073] Matt O'Gorman * apps/app_senddtmf.c: we can't unlock a channel if we cant find it. - AnthonyL bug #8741 2007-01-08 18:21 +0000 [r50032] Joshua Colp * main/rtp.c: Disable the more intense packet2packet bridging until the bugs can be worked out. 2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson * channels/chan_sip.c: Issue #8677 - Handle failure of T.38 re-invite This is not a fix, but adding an error message to tell the admin that we have a bad configuration. We should not send T.38 re-invites to devices that can't handle it (with the current architecture where you have to hard-code t.38 support per device). To really fix this, we need to figure out a way to tell the incoming call that the re-invite failed, so we can signal failure on that end and go back to the original call. * channels/chan_sip.c: Issue #8524, support multiple via header values (tardieu) Thanks! * channels/chan_sip.c: We only need one forward declaration * channels/chan_sip.c: Issue 8735: Terminate state when extension is unavailable for subscription 2007-01-08 05:11 +0000 [r49890] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 lines Ensure we use the default refresh value of 60 if the remote server does not send one. (issue #8746 reported by maethor) ........ 2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming * configure, configure.ac: since we use AC_PATH_TOOL to find tools, we should use the results it provides for us (reported by Brian Capouch on the asterisk-dev list) 2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) | 2 lines If openstream fails, then we crash (Issue 8564) ........ * channels/chan_sip.c: Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747) 2007-01-06 00:24 +0000 [r49742] Jason Parker * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping byte of allocated memory! This looks like it may have been a chicken/egg scenario.. You had to call a cleanup func, because everything was allocated. Then since you had to call a cleanup func, you were forced to allocate - ie; strdup(""). 2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming * configure, acinclude.m4: one more time... * configure, acinclude.m4: proper fix for r49712 * configure, acinclude.m4: if --with-foo= is specific for a configure option, ensure that it is used for header file checking as well * main/manager.c: ast_func_read() needs a writable copy of the function name to be passed 2007-01-05 23:16 +0000 [r49705] Jason Parker * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and chan_zap also depend on zaptel. This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel. 2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming * main/manager.c: don't 'consume' the params list before we try to use it again * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c, main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, main/db.c, channels/chan_zap.c, channels/chan_sip.c, apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases 2007-01-05 22:14 +0000 [r49675] Joshua Colp * main/channel.c: Don't keep repeating the warning over and over when the end of the call is reached. (issue #8724 reported by xrg) 2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming * /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_iax2.c: Merged revisions 49635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly ........ * channels/chan_iax2.c: revert the dynamic_list insertion change... that was not the right thing to do * channels/chan_iax2.c: create the IAX2 processing threads as background threads so they will use smaller stacks when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it 2007-01-04 23:00 +0000 [r49568] Joshua Colp * channels/chan_iax2.c: It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries. 2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming * include/asterisk/threadstorage.h, main/asterisk.c, build_tools/cflags.xml, include/asterisk.h, main/Makefile, main/threadstorage.c (added), main/utils.c: add support for tracking thread-local-storage objects that exist via 'threadstorage' CLI commands 2007-01-04 22:28 +0000 [r49551] Joshua Colp * main/config.c: Only free comments and line buffer once we reach the first level. (issue #8678 reported by ssokol, fixed by anthonyl) 2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming * channels/iax2-parser.c, main/frame.c: don't mark these allocations as 'cache' allocations when caching has been disabled * channels/iax2-parser.c: if we're going to decrement the frame count when we free a frame, we should inrement it when we create one :-) * channels/iax2-parser.c, channels/iax2-parser.h, channels/chan_iax2.c: only do IAX2 frame caching for voice and video frames * main/frame.c: don't do frame header caching in the core if LOW_MEMORY is defined * channels/iax2-parser.c: don't define this type either if LOW_MEMORY is enabled 2007-01-04 18:11 +0000 [r49459] Matt O'Gorman * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 lines converted a lot of 256 to PATH_MAX and some white space fixes. ........ 2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode * codecs/Makefile: make building of codec_gsm against the system GSM library actually work 2007-01-04 16:50 +0000 [r49413] Matt O'Gorman * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 lines good catch russell sorry i missed that. fix magic number with proper sizeof ........ 2007-01-04 04:33 +0000 [r49388] Russell Bryant * funcs/func_realtime.c: Fix the REALTIME() dialplan function. ast_build_string() advances the string pointer to the position to begin the next write into the buffer. So, this pointer can not be used to copy the contents of the string later. The beginning of the buffer must be saved. Interestingly enough, this code could not have ever worked. (Pointed out by Sebb on IRC, thanks!) 2007-01-03 23:32 +0000 [r49355] Matt O'Gorman * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 lines When using ODBC_STORAGE VoicemailMain doesn't create the subdirectories for a mailbox such as the INBOX directory. this patch solves that problem, was written by anthony be-125 ........ 2007-01-03 09:06 +0000 [r49313] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn_config.c, doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample: Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ 2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming * Makefile, Makefile.rules: various Makefile improvements to get chan_vpb (and any other C++ modules) to build properly 2007-01-03 01:19 +0000 [r49259] Joshua Colp * channels/chan_iax2.c: Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed. 2007-01-02 22:30 +0000 [r49237] Steve Murphy * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, pbx/ael/ael.flex: This is a slight modification to Josh's edits for #8579; both files edited were the produced by flex; so the source files need to be changed instead, and the generated files regenerated. 2007-01-02 19:58 +0000 [r49212] Olle Johansson * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always enabled at that point in the code 2007-01-02 17:33 +0000 [r49189] Jason Parker * main/pbx.c: Allow fractions of a second in the Wait() application, like it says it allows. 2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming * channels/chan_zap.c: remove comment that is unrelated to this function 2007-01-02 12:08 +0000 [r49145] Olle Johansson * configs/features.conf.sample: Adding note on effect of applicationmap features on re-invites 2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, configure.ac, codecs/codec_zap.c: check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version) * Makefile: use a simpler (and portable) method to ensure that menuselect is built as a host binary * Makefile: revert this change until a better solution can be found... 'env -i' was not being used properly, but even when changed to do so, this process fails during cross-compilation because the menuselect build still sees 'CC' as set to the cross-compiler 2007-01-01 20:14 +0000 [r49096] Olle Johansson * channels/chan_sip.c: remove incomplete implementation of dnsmgr. Let's fix this in trunk. 2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp * pbx/pbx_config.c: IAX has been deprecated for quite some time so we had better use IAX2 when creating the dial string for users. (issue #8697 reported by ssokol) * channels/chan_zap.c: Use asprintf to build the channel names instead of custom function. I believe the custom function is doing some things that are not portable across all implementations. (issue #8570 reported by hterag & issue #8692 reported by nicolasg) * main/rtp.c: If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte) * channels/chan_iax2.c: Initialize the packet queue in load_module instead of just declaring the list with the default value. (issue #8695 reported by ssokol) 2006-12-30 00:40 +0000 [r49061] Steve Murphy * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have comma args converted to vertical bars. I hope this change does little harm. 2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming * /: put this value into the correct property * /, BUGS: Merged revisions 49045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) | 2 lines location of the bug posting guidelines has changed ........ * sample.call: simple commit to test CIA integration 2006-12-28 21:26 +0000 [r49032-49035] Jason Parker * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me * main/http.c: saw this in passing... fix a small typo 2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming * sounds/Makefile: new versions of sounds 2006-12-28 19:52 +0000 [r49024] Jason Parker * main/http.c: make the uris_lock a rwlock instead of a mutex lock - needs to be forward ported to trunk 2006-12-28 19:43 +0000 [r49022] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h: Backport support for read/write locks. 2006-12-28 19:21 +0000 [r49020] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c, pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, pbx/ael/ael.tab.h, utils/ael_main.c: removed as in trunk from the ael stuff. Also, threw in a minor fix to frame.c to avoid build-killing compiler warnings. 2006-12-27 22:28 +0000 [r49009] Joshua Colp * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not available when LOW_MEMORY is used and things are being built in the utils directory, so we need to resort to the old method of strncpy. (issue #8579 reported by mottano) 2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c, main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c, main/http.c, main/logger.c: since these variables all have static duration, none of them need initializers (they default to zero anyway) * include/asterisk/options.h, main/asterisk.c, main/file.c: move extern declaration for this option to a header file where it belongs provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value 2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson * channels/chan_sip.c: Only include acl.h and lock.h once * channels/chan_sip.c: Only set rfc2833compensate flag once (handle_common_options) * channels/chan_sip.c: - Remove checking for T38 options twice. Keeping them in handle_common_options 2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming * channels/chan_sip.c: make the option actually match the documentation * channels/iax2-parser.c, include/asterisk/utils.h, include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks 2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more politically correct * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy cisco MWI support. Normally we try not to change our software for bugs in other devices. But in this case, the Cisco phones are so widespread so we try to implement a fix while waiting for a bugfix from Cisco. * channels/chan_sip.c: - Make sure handle_common_options return 1 when we found a common option - Move uncommon (only global) option away from handle_common_options Reported by rizzo. Thanks! * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before re-sending invite with auth. * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp. (rizzo, #8600) 2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp * apps/app_meetme.c: Get rid of a needless memory allocation and only create a conference structure in find_conf_realtime if data was read from realtime. (issue #8669 reported by robl) * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang) * configure, configure.ac: Clean up autoconf file (gets rid of warnings seen when rebuilding configure) and rebuild configure. 2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | 6 lines Fix an error introduced by copying and pasting the handling of the >= operator for the MATH function. If a single equal sign was used as an operator, the function would treat it is as if it were the >= operator. Now, it properly handles it as an invalid operator. (issue #8665, patch by tempest1) ........ * channels/chan_oss.c: Fix a typo in an error message that indicated that the MGCP channel type could not be registered, instead of the correct type, OSS. * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | 3 lines Check for the proper return value on an error in a call to mmap(). This was reported by Andy Wang on the asterisk-dev list. Thanks! ........ * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines Remove a couple of misplaced dots in log messages. This was reported by Andrea Spadaccini on the asterisk-dev mailing list. ........ * main/http.c: Implement locking for the list of URI handlers to make it thread-safe. 2006-12-23 Kevin P. Fleming * Asterisk 1.4.0 released. 2006-12-22 22:33 +0000 [r48870-48906] Jason Parker * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia 2006-12-21 20:26 +0000 [r48783] Joshua Colp * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 lines Add new silence sound files to the spec for Redhat. (issue #8652 reported by alvaro_palma_aste) ........ 2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage builds. * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so it is then passed to the IMAP store file function. (issue #8614 reported by punknow) * doc/snmp.txt: find is not the same as bind when it comes to documentation. (issue #8626 reported by johann8384) 2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming * channels/Makefile: suppress compiler warnings in this module until it can be improved 2006-12-19 21:12 +0000 [r48585] Joshua Colp * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 lines Free localuser structure when we fail to dial (issue #8612 reported by rizzo) ........ 2006-12-19 21:03 +0000 [r48583] Luigi Rizzo * apps/app_sms.c: fix a bogus datalen in the frames generated by app_sms (causing noisy output if you listen to the output!) This affects trunk as well, whereas 1.2 is ok. 2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable type for these unixODBC API calls, eliminating warnings on 64-bit platforms that use the 'new' 64-bit types for ODBC API calls 2006-12-19 03:46 +0000 [r48571] Joshua Colp * Makefile: Use env -i to start a fresh environment when going to build menuselect. This is more portable then using unset. (issue #8543 reported by jtodd) 2006-12-18 17:23 +0000 [r48566] Luigi Rizzo * include/asterisk/channel.h: unbreak the macro used for incrementing the frame counters. I don't know when the bug was introduced, but with the typical usage c->fin = FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects trunk as well (fix coming). 2006-12-18 17:15 +0000 [r48564] Joshua Colp * channels/chan_iax2.c: Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia) 2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, utils/ael_main.c: remove some now-unnecessary explicit includes of autoconfig.h clean up per-file dependencies during 'make clean' * build_tools/prep_tarball: need an additional argument here to make the downloads actually occur * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep these calls from thinking they have multiple arguments * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, main, codecs/gsm, pbx, res, channels, codecs, utils, agi, main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: simplify dependency tracking system, using the compiler's built-in method for generating them, and only doing dependency tracking if developer mode is enabled via the configure script * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself) * funcs/func_curl.c: instead of initializing the curl library every time the CURL() function is invoked, do it only once per thread (this allows multiple calls to CURL() in the dialplan for a channel to run much more quickly, and also to re-use connections to the server) (thanks to JerJer for frequently complaining about this performance problem) 2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp * main/rtp.c: Turn payload_lock into bridge_lock and make it encompass all RTP structure contents that may relate to bridge information, including who we are bridged to. * channels/chan_iax2.c: Hold call structure lock in places where a qualify or peer action can destroy it. * channels/chan_iax2.c: Lock network retransmission queue in all places that it is used. 2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported from 1.2) * channels/chan_sip.c: Update to latest IANA spec 2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp * channels/chan_iax2.c: Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted. * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman) * main/rtp.c: Remove direct RTCP bridging. I've come to the conclusion that we should handle this through the core and not just forward it on. Should solve a few bugs. 2006-12-12 Kevin P. Fleming * Asterisk 1.4.0-beta4 released. 2006-12-12 04:13 +0000 [r48401] Joshua Colp * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This is the way it should have been done. 2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman * sounds/Makefile: new sounds package with 100% more silence * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) | 4 lines app_externalivr needs a real silence file, and additional changes to add silence files into core instead of extra patch provided by bug 8177 with minor additions. ........ 2006-12-11 21:31 +0000 [r48391] Joshua Colp * apps/app_voicemail.c: Return non-existant callerid handling to that which it was before. In 1.4 and trunk callerid can be allocated but not have any contents so we have to use ast_strlen_zero before passing it to the relevant functions. (issue #8567 reported by pabelanger) 2006-12-11 05:37 +0000 [r48382] Tilghman Lesher * funcs/func_strings.c: STRFTIME() does not actually require an argument (issue 8540) 2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp * main/rtp.c: Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one. * apps/app_meetme.c: Use the correct API call to say a device state changed. (Yes, I'm a nub.) * apps/app_meetme.c: Don't access the conference structure after it has been freed. 2006-12-11 00:47 +0000 [r48375] Tilghman Lesher * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines When doing a fork() and exec(), two problems existed (Issue 8086): 1) Ignored signals stayed ignored after the exec(). 2) Signals could possibly fire between the fork() and exec(), causing Asterisk signal handlers within the child to execute, which caused nasty race conditions. ........ 2006-12-10 03:04 +0000 [r48372] Steve Murphy * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels). ........ 2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant * channels/chan_iax2.c: Use locking when accessing the registrations list. This list is not actually used very often, so the likelihood of there being a problem is pretty small, but still possible. For example, if the CLI command to list the registrations was called at the same time that a reload was occurring and the registrations list was getting destroyed and rebuilt, a crash could occur. In passing, go ahead and convert this list to use the linked list macros. * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use locking when accessing the registrations list. This list is not actually used very often, so the likelihood of there being a problem is pretty small, but still possible. For example, if the CLI command to list the registrations was called at the same time that a reload was occurring and the registrations list was getting destroyed and rebuilt, a crash could occur. ........ 2006-12-07 18:17 +0000 [r48357] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 Dec 2006) | 3 lines Ensure that the file position is not incremented beyond the total number of files available for playback. (issue #8539, ulogic) ........ 2006-12-07 15:33 +0000 [r48349] Steve Murphy * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that killed bug 8423 -- OriginateSuccess and OriginateError incomplete channel name. May it rest in peace. 2006-12-06 16:25 +0000 [r48326] Olle Johansson * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being retransmitted to Asterisk 2006-12-06 16:15 +0000 [r48323] Russell Bryant * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option in the sample configuration file. (issue #8526, arkadia) ........ 2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson * /, channels/chan_sip.c: Don't send Contact on MESSAGE 2006-12-05 20:42 +0000 [r48279] Jason Parker * configure.ac: Fix curl version number testing to be much more friendly to non-bash shells. Issue 8508, patch by me. This *SHOULD* be POSIX compliant now.. 2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson * channels/chan_sip.c: Merging the invitestate-1.4 branch after successful testing. Will check if I can solve this with less changes in 1.2. * configs/sip.conf.sample: Add missing s from another repository. (thanks jcmoore!) * configs/sip.conf.sample: Updating sip.conf.sample with information about T38 not working when chan_local or chan_agent is involved in the call. I don't know how big a fix that would be to solve, but this is the current state of affairs. (Chan_sip currently checks if the other side of the bridge has a SIP tech. We could/should implement another check, possibly for udptl_write or some flag in the ast_channel structure). 2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher * apps/app_voicemail.c: Oops, forgot to release the odbc handle * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines If the recording in the database is too large, it will fail to retrieve with an mmap error. Not too sure why this doesn't happen when we put it in the database, also, but since that doesn't seem to be broken, I'm not going to fix it (at least until someone reports it). Solution is to ask for the file in smaller chunks. (Bug 8385) ........ 2006-12-04 21:48 +0000 [r48237-48248] Jason Parker * apps/app_voicemail.c: Fix an issue which didn't allow unavail/greet/busy/etc messages from being saved into ODBC (and probably IMAP). * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell | 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert change from 8016 - this breaks other stuff... Needs further review. Tip: When you've reported a bug about something and somebody has put up a patch for it.. It's not a good idea to open a completely new bug and say that something is broken because of the patch in the other bug - PLEASE mention something in the bug where the patch was actually created. ........ * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell | 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an issue where a message isn't saved correctly when using ODBC storage and reviewing a message. Issue 8016 - patch by sokhapkin. ........ 2006-12-04 18:16 +0000 [r48234] Joshua Colp * /: Blocked revisions 48233 via svnmerge ........ r48233 | file | 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the generic bridge tells us not to retry, and we have a frame to spit out then break the bridge. Props to markit in #asterisk-bugs for bringing this up. ........ 2006-12-04 17:54 +0000 [r48228-48230] Jason Parker * configs/voicemail.conf.sample: Add documentation to voicemail.conf.sample for ODBC storage. Issue 8499 - patch by blitzrage. * doc/snmp.txt: Attempt to document some of the dependencies that are needed for net-snmp Issue 8499 - initial patch by blitzrage. 2006-12-03 06:34 +0000 [r48223] Russell Bryant * sounds/Makefile: When "fetch" is in use, instead of "wget", --continue is not a valid option. (issue #8451) 2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson * channels/chan_sip.c: - Removing one of two pieces of code to handle 481 response on INVITE - Move handling of REFER response to handle_response_refer() * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax transmission happens - Encapsulate RTP timers in the rtp structure so we have one for video and one for audio The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send something that video phones support in the RTP stream. I now this is a big architectual change at this stage for 1.4, but decided it was needed to avoid future bug reports. - Document the RTP NAT keepalive option in sip.conf.sample Issue 7679 in the bug tracker. Please test. 2006-12-02 03:50 +0000 [r48195] Russell Bryant * include/asterisk/utils.h: Backport the comment containing the warning regarding the limitations on the usage of this function. It is thread safe, but not technically reentrant. 2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106) ........ 2006-12-01 23:16 +0000 [r48190] Russell Bryant * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: FreeBSD 6.1 does not include wget by default. However, it has fetch which will work just fine for our purposes of downloading the sounds packages. So, check for both wget and fetch and the configure script and use what was found to download them. If neither one was found, and sound packages are selected that must be downloaded, the install process will print out an informative error message indicating the situation. Also, fix a couple places where "make" was hard coded into some output messages by replacing them with the $(MAKE) variable. (issue #8451, initial patch by pabelanger, with additional modifications by me) 2006-12-01 20:25 +0000 [r48184-48186] Jason Parker * configs/extensions.conf.sample, /: Merged revisions 48183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines Fix a small typo - issue 8848, reported by pabelanger ........ 2006-12-01 19:38 +0000 [r48179] Tilghman Lesher * main/cli.c: Double-unlock error (reported by blitzrage on IRC) 2006-12-01 17:41 +0000 [r48177] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states - Remove support for T.38 early media, since it's impossible. (Two patches in one - extra friday evening offer due to being off line from svn today... :-) 2006-11-30 21:18 +0000 [r48168] Joshua Colp * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) 2006-11-30 20:51 +0000 [r48166] Olle Johansson * /, channels/chan_sip.c: Issue 8319 - change noncecount before using it. 2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp * /: Blocked revisions 48161 via svnmerge ........ r48161 | file | 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel driver. (issue #8390 reported by hselasky) ........ * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex) ........ * /, res/res_features.c: Merged revisions 48154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155) ........ * main/cdr.c, /: Merged revisions 48151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader) ........ * /: Blocked revisions 48146 via svnmerge ........ r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia) ........ * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ 2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson * doc/manager.txt: Explain status reports and make codefreeze more happy :-) * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line in the REGISTER request. Imported from 1.2. 2006-11-29 21:05 +0000 [r48115] Joshua Colp * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey) 2006-11-29 19:56 +0000 [r48113] Olle Johansson * configs/sip.conf.sample: Explain the use device status system implemented in SIP for subscriptions, queues and manager a bit better. Like in 1.2, you will get more detailed information if you set a call limit for a device. When the call limit is reached, the status system will report a device as busy. For queues, setting a call limit per SIP device is propably a requirement. In most cases, it will work much better if you only use type=peer and not type=friend. We might decide to backport the new setting from trunk to apply all call limits to the peer part of a friend only. 2006-11-29 16:50 +0000 [r48107] Joshua Colp * main/rtp.c, /: Merged revisions 48106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3) ........ 2006-11-29 08:03 +0000 [r48105] Olle Johansson * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. 2006-11-29 04:26 +0000 [r48101] Joshua Colp * apps/app_voicemail.c: Don't crash if the mailstream was not created. 2006-11-28 18:26 +0000 [r48095] Jason Parker * Makefile: Export several more variables in top level Makefile. Inspired by issue 8438. 2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3) ........ * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 lines Use the proper function to get the new message count instead of always using the filesystem. (issue #8421 reported by slimey) ........ 2006-11-27 17:20 +0000 [r48049] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) ........ 2006-11-27 17:17 +0000 [r48046] Russell Bryant * main/manager.c: Remove a couple of unused variables (issue #8380, casper) 2006-11-27 15:32 +0000 [r48038] Joshua Colp * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 lines Do not reference the freed outgoing structure in the debug message. (issue #8425 reported by arkadia) ........ 2006-11-27 06:41 +0000 [r48031] Olle Johansson * channels/chan_sip.c: Change logging message 2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy * funcs/func_cdr.c: might as well also document the raw values of the flag vars * /, funcs/func_cdr.c: A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it. 2006-11-25 09:28 +0000 [r48002] Olle Johansson * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 and future releases, you can disable subscription support totally or per peer in sip.conf with allowsubscribe = yes | no 2006-11-24 17:17 +0000 [r47992] Steve Murphy * main/translate.c: bug 8189 posted this fix for main/translate.c for PLC 2006-11-24 15:46 +0000 [r47989] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm ........ 2006-11-23 11:01 +0000 [r47959] Olle Johansson * /, channels/chan_sip.c: Don't allocate unused variable. 2006-11-22 21:47 +0000 [r47944] Joshua Colp * main/rtp.c: Video will never reach Packet2Packet bridging and can do more harm then good. 2006-11-21 17:32 +0000 [r47897] Joshua Colp * main/rtp.c: If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate) 2006-11-21 15:20 +0000 [r47892] Olle Johansson * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 provisional response. Let's not treat that as early media. (discovered at the AVTF meeting in Paris). 2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher * apps/app_voicemail.c: Oops, merge missed release of odbc object * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) | 2 lines Failing to trap -1 error from mmap causes segfault (Issue 8385) ........ 2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp * main/frame.c, /: Merged revisions 47859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs) ........ * /: Blocked revisions 47855 via svnmerge ........ r47855 | file | 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free history items at the end of use of the temporary SIP pvt structure. (issue #8383 reported by benh) ........ * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists. * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c, include/asterisk/channel.h: Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings) 2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson * configs/sip.conf.sample: Explain properly how videosupport works. Committ from Asterisk Video Task Force meeting in Paris! * /, channels/chan_sip.c: Make sure we destroy scheduled items and not use them ever again after destruction (rizzo) 2006-11-18 17:59 +0000 [r47823] Luigi Rizzo * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header contains angle brackets (the bug was only in a corner case where the < was right after the opening quote, and the fix is trivial). 2006-11-16 23:19 +0000 [r47781-47782] Jason Parker * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially pointed out by mrobinson. * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell | 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a couple of typos in applications.. Initially spotted by mrobinson. ........ 2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming * /, doc/billing.txt: update documentation regarding IAX2 transfers and CDRs Merged revisions 47776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) | 2 lines update clearly wrong documentation regarding cdr_custom ........ 2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp * channels/chan_sip.c: Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath) * /: Blocked revisions 47761 via svnmerge ........ r47761 | file | 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for the header file specifically in all cases, not just the existence of the directory. (issue #8358 reported by mrness) ........ 2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming * configure, configure.ac: check for pre-1.4 versions of Zaptel and abort the configure script if found with an appropriate error message 2006-11-16 19:24 +0000 [r47755] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD notification optional, in order to avoid a lot of extra database lookups for all those realtime users out there. 2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp * channels/chan_local.c, /: Merged revisions 47750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell) ........ * apps/app_meetme.c: Don't unreference the SLA object if there is no SLA object in the devicestate callback. (issue #8354 reported by loloski) 2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup * UPGRADE.txt: Warn users about change in canreinvite * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never authenticated (according to the RFC) - Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. 2006-11-15 22:31 +0000 [r47712] Joshua Colp * channels/chan_local.c, /: Merged revisions 47711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me) ........ 2006-11-15 21:56 +0000 [r47709] Tilghman Lesher * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL 2006-11-15 21:33 +0000 [r47707] Joshua Colp * main/channel.c: We need to ensure timelimit stuff is included as well so warnings get played. (issue #8050 reported by KNK) 2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming * main/file.c: don't try to call fclose() if fopen() failed 2006-11-15 20:31 +0000 [r47698] Olle Johansson * channels/chan_sip.c: - Improve SIP history - Never send reply to ACK (again...) 2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) | 4 lines ensure that message duration is included in email notifications for forwarded messages (BE-96, fix by me after corydon used his clue-bat on me) ensure that duration in the message metadata is updated if prepending is done during forwarding (related to BE-96) remove prototype for API call that does not exist ........ * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 Nov 2006) | 2 lines clear the category's variable tail pointer as well when variables are detached from it ........ r47688 | kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 lines when appending a list of variable to a category, ensure the tail pointer points to the last variable in the list ........ r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) | 2 lines when re-writing the config file, don't repeat the path if it hasn't changed ........ * main/config.c, /: Merged revisions 47682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) | 2 lines ouch... don't use printf, use ast_log/ast_verbose ........ 2006-11-15 17:46 +0000 [r47672] Luigi Rizzo * main/cli.c: fix longest match search in find_cli. Trunk already fixed. 1.2 not affected (well, i have no idea, the code is totally different there). 2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson * /, channels/chan_sip.c: Send error message when we can't allocate SIP dialog, possibly due to limitation of file descriptors. (imported from 1.2) 2006-11-15 04:45 +0000 [r47645] Joshua Colp * main/rtp.c: If NAT detection is turned on or already detected then say NAT is active when setting the remote RTP peer when doing early bridging. (issue #8365 reported by marcelbarbulescu) 2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming * main/term.c: more formatting cleanup, and avoid running off the end of the string 2006-11-15 00:14 +0000 [r47639] Joshua Colp * main/rtp.c: Turn notice about unknown RTCP packet type into a debug message instead. 2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit platforms (this variable is an 'int' anyway, comparing it to 'signed long' is not useful) 2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 lines Update copyright information in the ADSI logo blob. ........ * channels/chan_sip.c: Only keep the video RTP structure around if 1. Video support is enabled and 2. A video codec is enabled on the dialog * funcs/func_uri.c: Small documentation clarification for URIENCODE. (issue #8294 reported by salaud) 2006-11-14 18:54 +0000 [r47621] Tilghman Lesher * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ prefix did not completely transition app_voicemail when ODBC_STORAGE is used (reported on IRC by caio1982, not in bugtracker) 2006-11-14 16:45 +0000 [r47617] Joshua Colp * apps/app_amd.c: Use LOG_DEBUG to print out the indication that app_amd is using default settings instead of using LOG_NOTICE. This stops needless logging of this information under normal circumstances. (issue #8361 reported by Seb7) 2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson * channels/chan_sip.c: Update documentation to fit the implementation... * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in retransmission system if it's an OPTION packet from peerpoke 2006-11-13 21:28 +0000 [r47584] Joshua Colp * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 lines Initialize global pointers for connection and result to NULL. (issue #8356 reported by james) ........ 2006-11-13 20:20 +0000 [r47581] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) | 2 lines Having more than 255 old messages caused corruption in the new/old count ........ 2006-11-13 19:15 +0000 [r47576] Steve Murphy * main/config.c: This solves bug 8342, whereby a crash occurs under certain circumstances while reading a config file with comments-- a call to CB_ADD shouldn't happen if withcomments is zero 2006-11-13 19:11 +0000 [r47573] Tilghman Lesher * main/cli.c, channels/chan_sip.c: Re-enable old deprecated commands 2006-11-13 19:10 +0000 [r47572] Olle Johansson * /, channels/chan_sip.c: - Don't reply to INVITE already replied to when we get BYE - Declare errmsg as int. Oops. 2006-11-13 18:18 +0000 [r47564] Steve Murphy * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing the messed if, but we all forgot to update the regressions. Until now. 2006-11-13 17:13 +0000 [r47553] Steve Murphy * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being found... just confuses users 2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 lines When sending an SMS with a user data header properly set the UDH flag in the first byte. (issue #8347 reported by hoffmeis) ........ * main/cli.c: Free full command string upon unregistering of CLI command. Backported from revision 47536 from rizzo. 2006-11-13 16:00 +0000 [r47540] Olle Johansson * channels/chan_sip.c: Only produce error message about sip history once 2006-11-13 05:48 +0000 [r47527] Russell Bryant * configure, acinclude.m4: AC_PROG_SED is included in autoconf 2.60, but apparently it is not included in 2.59. So, to maintain compatability with 2.59 since it is a small change, copy this macro into acinclude.m4 and rename it to AST_PROG_SED. (issue #8345) 2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) | 2 lines If the execute fails a second time, make sure that we don't pass back a stale handle ........ * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) | 2 lines Don't play dialtone if the seizing the channel fails (Bug 7754) ........ 2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks DEA!!!) * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is UDPTL in sdp... * channels/chan_sip.c: - Don't destroy SIP dialog because of a failed T.38 re-invite. Wait for a bye. Final response to a re-invite does not mean that the session dies, only that the re-invite fails. - Keep RTP active during processing of T.38 re-invite. If the re-invite fails, RTP needs to remain as before the re-invite. Issue 8338 - darren1713. Please test. * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp -Add some comments to t.38 code 2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | 4 lines Only do the check to determine whether the channel calling this function is an IAX2 channel when getting the IP address using the special argument, CURRENTCHANNEL. (issue #8341, jcovert) ........ * Makefile: Add the target "menuconfig" as an alias for the "menuselect" target. This is just a favor to users so that if you accidentally type "make menuconfig" instead of "make menuselect", it still works. (inspired by a comment on IRC from wangster calling me an "especially devious asterisk developer" for having it be menuselect instead of menuconfig. :) ) * main/term.c: Tweak the formatting of this new function to better conform to coding guidelines. 2006-11-11 02:04 +0000 [r47490] Matt O'Gorman * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo safe output! 2006-11-10 22:23 +0000 [r47480] Matt Frederickson * channels/chan_zap.c: Make sure we don't use 32 bits when we only need one bit. 2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson * channels/chan_sip.c: ...and make sure that the dialog is destroyed, even if we don't get any answer on the bye... This is the channel that remains dead after the SIP transfer * channels/chan_sip.c: Add debug output while trying to trace bug in bug report * channels/chan_sip.c: Make sure we destroy dialog... * /, channels/chan_sip.c: Small cleanup of handle_request_invite() - imported from 1.2 with changes 2006-11-10 19:47 +0000 [r47462] Matt Frederickson * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide callerid name for switches that bork on it. 2006-11-10 18:56 +0000 [r47454] Olle Johansson * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart SDP (alphaque) 2006-11-10 17:13 +0000 [r47444] Luigi Rizzo * build_tools/prep_moduledeps: grep -m is not available on BSD, so use head -1 instead 2006-11-10 16:53 +0000 [r47437] Joshua Colp * apps/app_chanspy.c: Only split up extension and context if a value exists. (issue #8332 reported by loloski) 2006-11-10 16:51 +0000 [r47436] Tilghman Lesher * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_iax2.c: Discussion of these CLI changes resulted in more consistency (Bug 8236) 2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then removing them should be LOG_NOTICE, not LOG_DEBUG * apps/app_queue.c: reflect addition/removal of dynamic queue members in queue_log, so that people using dialplan replacement for AgentCallbackLogin can still track login/logout (issue #7736, reported/patched by whoiswes but this commit was written by me and covers all three paths for AQM/RQM) 2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson * channels/chan_sip.c: Rip out half implementation of 491 response support, since it wasn't implemented properly and caused memory leaks in the case of us getting 491's, which Asterisk actually sends... Since it is a bit too complicated to fix this, I'll rip it out of 1.4 and put it on the to-do-list for future releases. Now, we handle this as congestion, which it really is. Issue #8331 * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. Thanks fenlander! 2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp * channels/chan_h323.c: Fix building of chan_h323 by completeing some structure definitions. (issue #8327 reported by Mithraen) * apps/app_voicemail.c: Do conversion in a more easier to read and working way for \r, \n, and \t. (issue #8324 reported by johnlange) 2006-11-09 21:26 +0000 [r47391] Russell Bryant * apps/app_voicemail.c, channels/chan_zap.c, build_tools/prep_moduledeps: Work around an issue that caused menuselect to display a bogus description for app_voicemail and chan_zap. These modules use some preprocessor directives to determine what it will report to Asterisk as its description. However, the way we extract this information from the source files for menuselect is not smart enough to figure this out. (issue #8326, #8328) 2006-11-09 16:53 +0000 [r47380] Joshua Colp * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods. ........ 2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant * main/cli.c: fix tab completion for "core debug channel" and "core no debug channel" * main/cli.c: Fix "core show channel". Also, fix tab completion for both "core show channel" and "core show channels". * main/cli.c: Fix "core debug channel ". I guess someone needs to go through and audit every CLI command that changed number of arguments ... * main/asterisk.c: revert the previous change, which actually modified the deprecated command, "show profile". Now, actually apply the change to "core show profile". * main/asterisk.c: Fix argument parsing for the "core show profile" CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) * main/cli.c: Fix another CLI command, "core show uptime" ... (issue #8323, reported by johnlange, fixed by myself) * main/asterisk.c: fix "core show version" to reflect the new number of arguments for this CLI command (issue #8316, kshumard) 2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy * main/channel.c: This update fixes 7531 * channels/chan_skinny.c: Committed in behalf of 8190. 2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming * main/frame.c: the battle over CLI command formats has broken stuff... * channels/chan_sip.c: add simple fix for SDP to report proper sample rate for G.722 media sessions 2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant * utils/streamplayer.c: I occasionally get email from users that are trying to figure out what this does, or due to some misunderstanding as to what it is supposed to do, can't get it to work. So, I have added some text here to hopefully explain what this application does and does not do. * channels/chan_gtalk.c: Make this module build again * configure, configure.ac, acinclude.m4: Copy the macros from libtool.m4 to our own acinclude.m4 such that libtool is no longer required to be installed to be able to generated the configure script. 2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) 2006-11-07 23:46 +0000 [r47303] Steve Murphy * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, include/asterisk/stringfields.h, apps/app_voicemail.c, main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life. 2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp * channels/chan_local.c: Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls) 2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming * configs/musiconhold.conf.sample: clean up sample config, and make native file playback the more obvious default choice 2006-11-07 18:38 +0000 [r47275] Matt O'Gorman * apps/app_voicemail.c: large overhaul to voicemail imap support. Allows support for more imap servers, also a better implementation of several parts of the original work. patch provided by 8033 with major upgrades. 2006-11-07 17:30 +0000 [r47268] Olle Johansson * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of continue. 2006-11-07 13:13 +0000 [r47250] Olle Johansson * /, channels/chan_sip.c: Fixing the attack shield so it doesn't produce attacks... Issue 8265 - never reply to an ACK 2006-11-07 01:25 +0000 [r47239] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 Nov 2006) | 5 lines If random order is enabled for files mode music on hold, set a random initial position, instead of always starting at the first file, and doing the random operation only when switching to the next file. (bug reported by John Lange on the asterisk-dev mailing list) ........ 2006-11-04 18:32 +0000 [r47199] Olle Johansson * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and transfer from "john" Thank you! 2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant * main/cli.c: Fix another bug in "core set debug" ... * main/asterisk.c, main/cli.c: Really fix the "core set debug" and "core set verbose" CLI commands. * main/cli.c: fix the "atleast" option to the "core set verbose" and "core set debug" CLI commands 2006-11-03 23:17 +0000 [r47176] Steve Murphy * channels/chan_sip.c: This fix introduced via bug 8233 2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo * bootstrap.sh: align bootstrap.sh with the version in trunk (needs to be blocked as it is already in trunk) * configure.ac: add proper environment vars to detect modules on freebsd. (already applied to trunk so it needs to be blocked there) 2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236) * main/config.c, main/cli.c, main/channel.c, main/manager.c, channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, main/http.c, main/file.c, main/logger.c, main/image.c, res/res_indications.c, main/asterisk.c, res/res_odbc.c, channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, channels/chan_local.c, main/frame.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, res/res_crypto.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" 2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson * channels/chan_sip.c: Move check for codec translation to sip_call() instead of in add_sdp. No one bothers with the result of add_sdp anyway... Yet... * channels/chan_sip.c: Disable code for T38 over TCP and RTP since there's no trace of actual functionality for it :-) 2006-11-02 17:49 +0000 [r46965] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 Nov 2006) | 3 lines ignore files in a music on hold directory that begin with '.' (issue #8249, cboie) ........ 2006-11-02 17:17 +0000 [r46963] Nadi Sarrar * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix 2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming * channels/chan_sip.c: don't send INVITE when we have determined that we can't offer any audio formats due to lack of transcoding support (or incorrect configuration) 2006-11-02 16:06 +0000 [r46930] Joshua Colp * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 lines Repeat after me oej: I will at least make sure my code compiles before I commit it. ........ 2006-11-02 15:24 +0000 [r46901] Olle Johansson * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) 2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant * /, main/callerid.c: Add the missing call to free described in issue #8268. Also, add a bunch of missing calls to free in callerid_feed_jp(). * main/say.c: fix saying one hundred and two hundred in hebrew (issue #7810, eldadran) * Makefile, configure, codecs/gsm/Makefile, configure.ac, build_tools/strip_nonapi, makeopts.in: Fixes for cross-compilation on mips (issue #8058, ywalther, with some modifications) * aclocal.m4, build_tools/menuselect-deps.in, configure, build_tools/embed_modules.xml, configure.ac: Add a check in the configure script to determine whether ld is GNU ld or not. This is needed because module embedding only works for gnu ld. GNU ld is now listed as a dependency for all of the module embedding options in menuselect. (issue #8143) 2006-11-01 20:35 +0000 [r46822] Matt O'Gorman * channels/chan_gtalk.c: bind address support from bug 8164 2006-11-01 19:49 +0000 [r46802] Steve Murphy * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to accept longer strings or mass confusion and a lot of lost time is the result 2006-11-01 18:39 +0000 [r46780] Joshua Colp * main/Makefile: Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera. 2006-11-01 18:26 +0000 [r46778] Russell Bryant * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | 9 lines soxmix and Asterisk expect different file extensions for certain formats. This was already handled for the wav49 format. However, it was not handled for ulaw and alaw. I fixed this in such a way that using the alternate extensions for ulaw and alaw will only happen if we know we're calling soxmix, and not a custom script defined using the MONITOR_EXEC variable. The wav49 processing was left alone so that external scripts will see no behavior change. (issue #7550, reported by mnicholson, proposed patch by junky, committed fix is a bit different) ........ 2006-11-01 18:21 +0000 [r46775] Joshua Colp * channels/chan_iax2.c: It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine. 2006-11-01 17:48 +0000 [r46760] Steve Murphy * main/config.c: Cleanups suggested by Russell. 2006-11-01 16:39 +0000 [r46744] Russell Bryant * channels/chan_zap.c: Prevent an infinite loop when config processing gets to a jitterbuffer option 2006-10-31 22:02 +0000 [r46716] Jason Parker * main/translate.c: Fix "core show translation" output. Issue #8243, patch by Damin. 2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming * include/asterisk/translate.h, main/translate.c: add an API so that translators can activate/deactivate themselves when needed * include/asterisk/translate.h, main/translate.c: revert changes that were the wrong way to address this... proper fix coming * main/translate.c: let's set the seen flag early enough to actually make a difference... * include/asterisk/translate.h, main/translate.c: don't re-do setup operations for translators that can dynamically register themselves 2006-10-31 15:49 +0000 [r46663] Tilghman Lesher * /: Blocked revisions 46662 via svnmerge ........ r46662 | tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines Move thread-unsafe initializer to the module loading code; add the corresponding function to the module unload to fix a memory leak. ........ 2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue #8089 - Fix the ENUM support (picking one record by number). Thanks otmar! * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport when we're supposed to support ;rport. Issue #7473. * /, channels/chan_sip.c: If peer fails ACL check, fail peer at REGISTER * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! 2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the boot process to ensure it starts after stuff like MySQL (issue #8253, Alric) * /, main/utils.c: Merged revisions 46560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | 3 lines When handling the case where the hostname is just an IPV4 numeric address, be sure to set the address type. (issue #8247, alexr) ........ * /, res/res_agi.c: Merged revisions 46557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | 3 lines fix some copy/paste bugs in the checking of arguments for the "control stream file" AGI command (issue #8255, mnicholson) ........ * main/translate.c: Add a small tweak to the code that checks to see whether destination formats are translatable based on the source format. If we have already determined that there is no translation path in one direction, don't bother checking the other direction. 2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming * main/translate.c: when unregistering a translator, don't rebuild the translation matrix unless needed when filtering formats out of an offer, ensure we check for translation ability in both directions * include/asterisk/linkedlists.h: ensure that items removed from a list are always unlinked from the list (next pointer set to NULL) 2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp * configure, configure.ac: Don't explicitly link in crypt as it is not used on some platforms. * channels/chan_iax2.c: We need to lock the pvt structure during retransmission as another worker thread may be doing something as well. 2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, include/asterisk/doxyref.h, channels/chan_sip.c, main/ast_expr2f.c, include/asterisk/module.h, formats/format_ogg_vorbis.c, main/app.c, include/asterisk/channel.h, include/asterisk/lock.h, include/asterisk/frame.h: Issue #8246 - Doxygen fixes from kshumard. An extra big thankyou is given to everyone that contributes to doxygen! THANK YOU! * main/rtp.c, /: Bind RTCP to the same IP as RTP * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 redirects (imported from 1.2) * /, channels/chan_sip.c: Issue #7608 - Notifications sent with wrong content-type (imported from 1.2, modified) * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that was reported for trunk, but obviously exists in 1.4 too. * channels/chan_sip.c: Restoring the old logic, since working around it and fixing it seemed too complicated. - The SIP_OUTGOING flag indicates the direction of the last transaction in the dialog. - The initreq stores the last request in the dialog, the request that opened the latest transaction. Please now retry all the 1.4 bug reports with mixed to/from headers, tags etc in ACK, BYE, CANCEL. Thanks! * channels/chan_sip.c: Accepting a message twice may be misinterpreted... * channels/chan_sip.c: - 183 is not reliable message... - Error should not have SDP 2006-10-28 16:37 +0000 [r46377] Joshua Colp * utils/Makefile: Don't build muted on OpenBSD, it is not supported. 2006-10-27 19:03 +0000 [r46370] Russell Bryant * channels/chan_zap.c: move the copy of the default settings to the global settings back out of process_zap, so that they aren't overwritten when process_zap is called multiple times 2006-10-27 18:29 +0000 [r46367] Olle Johansson * contrib/asterisk-ng-doxygen: Put some doxygen pressure on Christian :-) 2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, res/res_musiconhold.c: We should always be using _exit() after a fork() or vfork() instead of exit(). This is because exit() does some extra cleanup which in some implementations of vfork(), for example, can actually modify the state of the parent process, causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We should always be using _exit() after a fork() or vfork() instead of exit(). This is because exit() does some extra cleanup which in some implementations of vfork(), for example, can actually modify the state of the parent process, causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) ........ * channels/chan_zap.c: Instead of iterating all of the options once to look for jitterbuffer options, and then again for everything else, move the processing of jitterbuffer options into the main loop so that there are no erroneous messages about ignoring unknown options. (issue #8226) 2006-10-27 10:03 +0000 [r46351-46353] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 46350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c ........ * channels/misdn_config.c: fixed not compile issue, which was just introduced * channels/misdn_config.c, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: Merged revisions 46176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session ........ 2006-10-26 17:57 +0000 [r46335-46340] Jason Parker * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 lines oops - somebody forgot to change this - long ago, probably. ........ * CHANGES: grammar check 2006-10-26 16:38 +0000 [r46331] Olle Johansson * CHANGES: Corrections to changes (Multiparking is not included) 2006-10-26 16:31 +0000 [r46329] Russell Bryant * main/translate.c: - If the source has no audio or no video portion, do not call powerof() to get the format index. - Don't run through the audio and video loops if there is no audio or video portion of the source If 0 is passed to powerof, it will return -1. This value of -1 was then being used as an array index in these loops, which caused a crash on some systems. Other than this issue, this code works as we expected it to. If a format is not in the source, and we have to translation path to it, it is not offered in the list of acceptable destination formats. (fixes issue #8231) 2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming * CHANGES: update to reflect G.722 addition 2006-10-26 04:18 +0000 [r46298] Russell Bryant * doc/backtrace.txt: update backtrace documentation to reflect changes in 1.4 (issue #8230, kshumard) 2006-10-26 01:37 +0000 [r46287] Mark Spencer * main/config.c, main/manager.c: Fix config comment code preservation code (thanks murf!) 2006-10-25 20:14 +0000 [r46276] Olle Johansson * channels/chan_sip.c: Old todo note - Don't add Contact header on BYE and Cancel 2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant * configure.ac: fix error output when checking for openh323 to refer to openh323 instead of pwlib (issue #8222, misaksen) 2006-10-25 19:16 +0000 [r46252] Olle Johansson * channels/chan_sip.c: Somewhat ugly code to try to fix issue #7608. Since the problem was not very well defined, the fix is a bit fuzzy too... Thanks to Luigi for accidentally spotting the possible problem! 2006-10-25 19:08 +0000 [r46249] Russell Bryant * apps/app_queue.c: update warning message to include "agi" option (issue #8225, jmls) 2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming * sounds/Makefile: use 1.4.3 extra sounds with corrected silence files * sounds/sounds.xml, sounds/Makefile: add support for prebuilt G.722 prompts and music on hold files 2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson * channels/chan_sip.c: show settings doesn't produce a list of similar objects, it should stay a "show" 2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, cdr/cdr_custom.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, apps/app_voicemail.c, channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, res/res_adsi.c, main/utils.c, apps/app_ices.c, pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c: apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process 2006-10-25 04:58 +0000 [r46165] Tilghman Lesher * main/pbx.c: WaitExten truncates decimals of times to wait, instead of accepting them (Bug 8208) 2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, channels/chan_h323.c, channels/chan_iax2.c, include/asterisk/frame.h: add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen) * channels/chan_sip.c, main/translate.c: code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable * main/translate.c: if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list 2006-10-24 20:30 +0000 [r46142] Mark Spencer * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, #8147) 2006-10-24 19:29 +0000 [r46130] Joshua Colp * channels/chan_iax2.c: We need to initialize our scheduler pthread condition... yes. 2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo * main/http.c: merge 45152 don't leak descriptors in http.c * channels/chan_sip.c: merge 45966 refer_to_domain potentially containing options * channels/chan_sip.c: merge 46026 improper checks on get_header() return values * channels/chan_sip.c: merge 46045 prevent NULL args to ast_strdupa() in chan_sip.c 2006-10-24 05:23 +0000 [r46093] Russell Bryant * Makefile: Restore the ability to remove the firmware directory without causing the installation to fail (issue #8111) 2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming * main/translate.c: ensure that the translation matrix is properly lock-protected every place it is used * include/asterisk/translate.h, main/translate.c: add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using * doc/imapstorage.txt: simplify and correct voicemail IMAP storage build instructions 2006-10-24 03:01 +0000 [r46078] Tilghman Lesher * main/channel.c: Pass through a frame if we don't know what it is, rather than trying to pass a NULL, which will segfault a channel driver (Bug 8149) 2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant * utils/muted.c, utils/ael_main.c: In muted.c, check the return value of strdup. In ael_main.c, check the return value of calloc. (issue #8157) In passing fix a few minor bugs in ael_main.c. The last argument to strncpy() was a hard-coded 100, where it should have been 99. I changed this to use sizeof() - 1. * apps/app_meetme.c: Fix the descriptions of some of the MeetMeAdmin options (issue #8098, mflorell) * res/res_jabber.c: don't crash when an incoming message has no "from" (issue #8205, jmls) 2006-10-23 00:27 +0000 [r45928] Joshua Colp * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 lines Don't leak memory mmmk? ........ 2006-10-22 21:44 +0000 [r45916] Christian Richter * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and couldn't be initialized it would cause a segfault after 'reload'. Reported by Drew/Matt thx. ........ 2006-10-21 18:49 +0000 [r45818] Russell Bryant * res/res_monitor.c: Add a couple missing unregistrations of manager actions and remove duplicate unregistrations of applications. (issue #8194, jmls) 2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp * main/loader.c: Don't use promotion on Darwin because it doesn't seem to work quite right in all cases, this should solve the unresolved symbol issue people have been seeing. * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get installed in the proper location (reported on asterisk-dev mailing list) 2006-10-20 07:44 +0000 [r45741] Olle Johansson * channels/chan_sip.c: Let's understand SIP: - REFER can create dialog, Asterisk does not support it yet - NOTIFY can create dialog in Asterisk's implementation (voicemail) even though we don't support the server side of it. In this case, the standard is a side issue ;-) - Added extened functionality for unsupported methods (PING, PUBLISH) so we don't create PVT's for those either. Russellb needs to judge what to do with this in 1.2, but I think the current implementation n 1.2 is a bug since we're sending bad replies to NOTIFY and REFER outside of dialogs 2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp * res/res_jabber.c: Let's remember to unregister JabberStatus too (issue #8184 reported by jmls) * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct 2006) | 2 lines Respect language selection when seeing if the file exists (issue #8178 reported by mnicholson) ........ * channels/chan_sip.c: If the jitterbuffer is forced on then we can't partially bridge (reported by wangster on #asterisk-dev) 2006-10-19 00:59 +0000 [r45622] Russell Bryant * channels/chan_sip.c: Don't leak the actual thread-specific sip_pvt struct 2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming * channels/chan_sip.c: don't leak memory when a chan_sip thread is destroyed that has a thread-local temp_pvt allocated 2006-10-18 21:03 +0000 [r45595] Joshua Colp * main/asterisk.c: Don't modify things if we are using vfork as this is very bad and may cause unexpected behavior (issue #7970 reported by Nick Gavrikov) 2006-10-18 11:54 +0000 [r45517] Olle Johansson * channels/chan_sip.c: remove duplicate declarations 2006-10-18 04:09 +0000 [r45464] Luigi Rizzo * main/http.c: merge from trunk: move ast_variables_destroy() to a better place in handle_uri() to avoid leaking memory on non existing files. 2006-10-18 03:02 +0000 [r45452] Joshua Colp * main/rtp.c: Don't segfault if you're using a channel driver that doesn't turn RTCP on 2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant * main/channel.c: Don't attempt to access private data members of the pthread_mutex_t object, because this does not work on all linux systems. Instead, just access the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has DEBUG_THREADS on as well. (issue #8139, me) * configs/sip_notify.conf.sample: update entry to reboot a snom phone (issue #7850, pnlarsson) 2006-10-17 Kevin P. Fleming * Asterisk 1.4.0-beta3 released. 2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming * include/asterisk/stringfields.h, main/ast_expr2.c, main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: optimize the 'quick response' code a bit more... no more malloc() or memset() for each response expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed 2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp * channels/chan_sip.c: Don't create a "real" pvt structure for requests that shouldn't be able to create one. Instead use a temporary pvt and fill it with enough information so we can send a reply. 2006-10-17 17:39 +0000 [r45329] Olle Johansson * configs/sip.conf.sample: Adding information about Marks direct-RTP hack to the docs... 2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming * LICENSE: provide licensing language for IAXy firmware file 2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new directed pickup (BE-85). 2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for your support! * channels/chan_sip.c: Don't destroy dialog for unexpected REFER response... 2006-10-14 04:38 +0000 [r45143] Steve Murphy * funcs/func_rand.c: update the doc string for both AEL and extensions.conf users. 2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming * main/acl.c don't drop the entire permit/deny list when an attempt is made to add an invalid entry (BE-92) 2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp * res/res_speech.c: Clear the quiet flag too since we are restarting a recognition again (reported on -dev by Stephan Edelman) * res/res_speech.c: Check return value from engine in case of failure (ie: out of licenses) (reported on -dev mailing list) 2006-10-13 20:52 +0000 [r45103] Steve Murphy * pbx/ael/ael-test/ref.ael-vtest17 (added), pbx/ael/ael-test/ael-vtest17/extensions.ael (added), pbx/ael/ael-test/ael-vtest17 (added), pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in this release via these changes 2006-10-13 19:19 +0000 [r45088] Christian Richter * channels/chan_misdn.c: avoiding warning, fixing potential bug 2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, codecs/lpc10/synths.c, codecs/lpc10/preemp.c, codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, codecs/lpc10/lpcini.c, codecs/lpc10/random.c, codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, codecs/lpc10/analys.c, codecs/lpc10/onset.c, codecs/lpc10/energy.c, codecs/lpc10/deemp.c, codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, codecs/lpc10/median.c, codecs/lpc10/encode.c, codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, codecs/lpc10/invert.c: And file said... let the compiler warnings STOP! * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 reported by mnicholson) * apps/app_playback.c: Move say.conf existence check to do_say function since it is called from multiple places (issue #8144 reported by kshumard) 2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if we have multiple bindings (reported on asterisk-dev) 2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp * channels/chan_sip.c: Complete merging in RPID screen changes (issue #8101 reported by hristo, patch by oej in revision 44757) * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add the background refresh item back into the scheduler if enabled since it is deleted during reload. (issue #8142 reported by p_lindheimer) 2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming * configure, include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: use a configure script test for PMTU discovery control instead of just assuming it's available on Linux 2006-10-13 14:45 +0000 [r44994-45026] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some echocandisable issues when bridged. this caused a kernel panic sometimes.. also some minor formatting fixes * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause got a wrong isdn cause at RELEASE_COMPLETE 2006-10-12 22:07 +0000 [r44992] Luigi Rizzo * channels/chan_sip.c: merge formatting and minor code simplifications from trunk 2006-10-12 20:34 +0000 [r44982] Matt O'Gorman * channels/chan_gtalk.c: fix for bug 7764. 2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming * channels/chan_sip.c: we can only send one 'a=ptime' attribute per media session, not one for each format * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, main/utils.c: ensure that IAX2 and SIP sockets allow UDP fragmentation when running on Linux (thanks to Brian Candler on the asterisk-dev list for the tip) 2006-10-12 16:56 +0000 [r44945] Russell Bryant * main/manager.c: fix a silly typo in a comment that I saw while reading the commit list 2006-10-12 16:08 +0000 [r44942] Joshua Colp * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue #8135 reported by ssokol) 2006-10-12 12:55 +0000 [r44921] Nadi Sarrar * main/manager.c: append_event must be called while holding the session lock 2006-10-12 10:24 +0000 [r44911] Russell Bryant * res/res_jabber.c: change some debug output to use LOG_DEBUG instead of verbose output 2006-10-11 16:57 +0000 [r44888] Jason Parker * main/db1-ast/Makefile: These are already set by the parent Makefile.. There is no need to have this here (it doesn't actually work anyways). 2006-10-11 09:18 +0000 [r44854] Christian Richter * channels/misdn/isdn_lib.c: removed warning because of missing prototype declaration 2006-10-10 19:23 +0000 [r44830] Olle Johansson * channels/chan_sip.c: Do not set default/global values in the variable declaration, set it in reload_config() 2006-10-10 17:21 +0000 [r44819] Joshua Colp * channels/chan_sip.c: Move some stuff around so that a NOTIFY dialog won't hang around until the end of the world under certain circumstances 2006-10-10 16:44 +0000 [r44809] Paul Cadach * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: CHANNEL() function sometime mix parameter and value 2006-10-10 16:42 +0000 [r44808] Tilghman Lesher * funcs/func_logic.c: Lost of a bit of logic when this was simplified between 1.2 and 1.4 (Bug 8117) 2006-10-10 16:30 +0000 [r44806] Joshua Colp * channels/chan_sip.c: Bail out if we have no refer structure and we get a refer response 2006-10-10 16:21 +0000 [r44805] Luigi Rizzo * channels/chan_sip.c: more merge from trunk (comments and change a static function name) 2006-10-10 15:23 +0000 [r44788] Joshua Colp * channels/chan_sip.c: Only set DTMF information if an RTP structure exists 2006-10-10 13:50 +0000 [r44786] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added support of dynamically enabling hdlc on bchannels 2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo * channels/chan_sip.c: whitespace changes related to previous commit * channels/chan_sip.c: merge a few code simplifications that have gone into trunk during last week, to reduce differences between the two branches and make porting fixes easier. 2006-10-09 16:12 +0000 [r44764] Jason Parker * channels/chan_skinny.c: Fix a problem where phones that go "missing" never got unregistered. Issue #8067, reported by pj, patch by Anthony LaMantia (with minor whitespace modifications) 2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid the deadlock * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup (issue #8115 reported by vazir) 2006-10-08 14:14 +0000 [r44746] Luigi Rizzo * channels/chan_sip.c: do not dereference p if we know it is NULL 2006-10-07 14:39 +0000 [r44684] Paul Cadach * channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate caller's transfer capability too 2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo * channels/chan_sip.c: put common code in a function to avoid repetitions. * channels/chan_sip.c: remove hardwired usage of 5060, use DEFAULT_SIP_PORT instead * channels/chan_sip.c: option_debug checking before printing to debug channel. * channels/chan_sip.c: backport simplifications on sip_register, usage of ast_set2_flag(), and fixes to the handling of failed module loading. * channels/chan_sip.c: improve and document function get_in_brackets(), introducing a helper function find_closing_quote() of more general use. 2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming * include/asterisk/linkedlists.h: ensure that mutex locks inside list heads are initialized properly on platforms that require constructor initialization (issue #8029, patch from timrobbins) * CHANGES: remove Jingle as per mog 2006-10-06 21:08 +0000 [r44628] Joshua Colp * main/rtp.c: Remove the seqno check for RFC2833, the handler is smart enough to not need it. 2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming * CHANGES: various cleanups 2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp * main/rtp.c: When the sequence number rolls over then reset the recorded sequence number for DTMF (issue #8106 reported by bungalow) * main/file.c: Even more frames to treat as though the remote side disappeared (issue #8097 reported by eldadran) 2006-10-06 15:59 +0000 [r44567] Luigi Rizzo * main/manager.c, main/http.c: make sure sockets are blocking when they should be blocking. 2006-10-06 12:53 +0000 [r44559-44563] Christian Richter * channels/chan_misdn.c: fixed segfault which happens during hold/transfer action * channels/chan_misdn.c: if INFORMATION Message come with keypad instead of called party number, we just use the keypad as called party number. * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, removed a useless bc field, added setting of frame.delivery fields, some minor code cleanups 2006-10-05 19:57 +0000 [r44502] Joshua Colp * main/file.c: Treat busy control frames as hangup in the file streaming core (issue #8097 reported by eldadran) 2006-10-05 18:21 +0000 [r44488] Steve Murphy * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. Many thanks to Doug! 2006-10-05 18:01 +0000 [r44486] Joshua Colp * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite hanging by a thread if the other side is already setup with T.38 2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming * main/app.c: don't segfault when an argument without a close parenthesis is found stop parsing as soon as that situation occurs 2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy * CHANGES: I put the accumulated changes from the commit logs and inspection, into CHANGES. Hope everyone approves! * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so that's where muted should look for it, right? 2006-10-05 02:40 +0000 [r44450] Joshua Colp * channels/chan_sip.c: Don't totally bail out if T.38 was negotiated 2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming * channels/chan_sip.c: fix Polycom presence notification again 2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo * utils/Makefile: as far as i can tell astman only uses newt... * Makefile: put linker flags in ASTLDFLAGS where they belong 2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE requests add workaround for new Polycom firmware SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware) * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually work 2006-10-04 19:57 +0000 [r44380] Steve Murphy * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ael-test16/extensions.ael (added), pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the problems reported in bug 8090 2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, channels/chan_skinny.c, channels/chan_h323.c, main/http.c, channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, main/devicestate.c, main/utils.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update thread creation code a bit reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined 2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy * configs/muted.conf.sample: I've been meaning to add some explanation about muted... here it is * configs/manager.conf.sample: CLI reverbification update to this config file * apps/app_macro.c: In response to bug 7776, a Warning has been added to the doc string for Macro(). 2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming * main/asterisk.c, main/loader.c, main/term.c, Makefile, include/asterisk.h: ensure that local include files are always used avoid a duplicate function name (term_init()) 2006-10-03 22:35 +0000 [r44312] Matt O'Gorman * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing client without resource. 2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming * apps/app_queue.c: fix a logic error in my previous fix to the queue reload code 2006-10-03 18:42 +0000 [r44286] Paul Cadach * channels/h323/ast_h323.cxx: Change default presentation indicator to "user provided not screened" if octet 3a missed in CallingPartyNumber IE 2006-10-03 18:35 +0000 [r44284] Joshua Colp * channels/chan_sip.c: Use VideoSupport instead so it is considered a valid XML attribute name. (issue #8075 reported by renemendoza) 2006-10-03 18:30 +0000 [r44283] Paul Cadach * channels/h323/ast_h323.cxx: Fix preparation of type and presentation of calling number 2006-10-03 00:01 +0000 [r44240] Matt O'Gorman * doc/jingle.txt, channels/chan_jingle.c (removed), include/asterisk/jabber.h, configs/jingle.conf.sample (removed), res/res_jabber.c: updated res_jabber for even better component support, soon will be jep-0100 compliant. also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version. 2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp * channels/chan_sip.c: Change the fd on the I/O context in case it changed during the reload, which is indeed possible. (issue #7943 reported by eclubb) * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN instead of hardcoding the path for the error message (issue #7942 reported by eclubb) 2006-10-02 18:52 +0000 [r44186] Paul Cadach * configs/users.conf.sample, pbx/pbx_config.c: Missed part of userconf functionality for chan_h323 2006-10-02 17:25 +0000 [r44169] Joshua Colp * main/io.c: Shrink when current_ioc is unused. It is set to -1 when unused, not 0. (issue #7941 reported by eclubb) 2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach * doc/realtime.txt: Typo fix * channels/chan_h323.c: Optimization of oh323_indicate(): less locks - less problems, plus single exit point 2006-10-02 02:38 +0000 [r44146] Mark Spencer * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when you're not talking about a channel :) 2006-10-01 19:32 +0000 [r44135] Paul Cadach * channels/chan_h323.c: Do not simulate any audio tones if we got PROGRESS message 2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to be empty. The cause is that since ASTDATADIR is explicitly exported using "export ASTDATADIR" at the top of the Makefile, make no longer considers the variable "undefined", so the Makefile can't use ?= to set ASTDATADIR if not yet set. (issue #8063, reported by akohlsmith, fixed by me) * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" option in the sample queues.conf (issue #8065, adamg) 2006-10-01 15:01 +0000 [r44109] Luigi Rizzo * channels/chan_sip.c: sync with trunk - move variable declarations to the beginning of a block. 2006-09-30 19:20 +0000 [r44090] Paul Cadach * main/rtp.c: Allow one-way RTP streams (device->Asterisk) 2006-09-30 16:28 +0000 [r44080] Luigi Rizzo * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent build problems: - with AST_DEVMODE, building codecs/lpc10 fails because of lots of warnings, and the configure step in editline fails as well. Fix this by removing the -Werror in these steps. - on FreeBSD (but probably on other platforms as well), the final link of asterisk fails because AST_LIBS was not exported to the subdirs Makefiles. Add a proper fix in the top-level Makefile (a possible alternative way is to add "export AST_LIBS" near the beginning of the file). With this fix, i believe that some of the platform-specific conditionals in main/Makefile are redundant (because they should be already dealt with in the top level Makefile) but i don't have a platform to check. Merging to head will happen in a moment. 2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment of previous fix: Issue #7928 - Don't send both 404 and 503. Fix by phsultan with a small fix by me, myself or I. Thanks, Philippe! (This was caused by my changes to the transaction handling) * channels/chan_sip.c: Found some buggy SIP clients (phones Planet VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK message only (when remote party answers) but on RINGING message too, so when we send 200 OK message, we get unidentified ACK message (because INVITE acknowledged on RINGING message already), so 200 OK retransmits within its retransmission interval then call gets dropped. If someone else knows how to provide workaround for such cases, please, fix it in correct way. Thanks to ssh from #asteriskru for provide access to his box to study and fix this case. 2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming * agi, utils: ignore temporary files made by the Makefiles during a build * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, configure, build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, pbx/Makefile, res/Makefile, channels/Makefile: fix a few build system bugs, and convert Makefiles to be compatible with GNU make 3.80 2006-09-29 22:35 +0000 [r44053] Jason Parker * main/asterisk.c, main/cli.c: Fix a bug with the removal of 'atleast' argument to 'core verbose' and 'core debug'. Add that argument back in. 2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more carefully when no CallingNumber IE available * channels/h323/ast_h323.cxx: Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented) * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add includes * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly pass TON/PRESENTATION information - original H323Connection::SendSignalSetup() destroys Q.931 fields. 2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming * main/Makefile: yet another place where we were not using the correct CFLAGS by default * main/Makefile: missed one conversion to ASTCFLAGS 2006-09-29 18:30 +0000 [r44009] Paul Cadach * channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass TON/PRESENTATION information too 2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse CFLAGS and LDFLAGS for build of Asterisk components, because they are also then used for non-Asterisk components (like menuselect); use our own variables instead * configure, configure.ac: support --without-curl in configure script * Makefile.rules: another cross-compile fix * Makefile: a couple more environment settings that can't leak into the menuselect build * main/cli.c: proper fix for ast_group_t change * include/asterisk/lock.h: eliminate compiler warning when DEBUG_CHANNEL_LOCKS is enabled and users of this header file don't also include channel.h 2006-09-28 20:11 +0000 [r43944] Jason Parker * apps/app_queue.c: Fix incorrect argument order for member names, on persisted members. Issue 8047, patch by jmls. 2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp * apps/app_playback.c, res/res_monitor.c, include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, main/udptl.c, main/frame.c, funcs/func_timeout.c, channels/chan_sip.c, apps/app_festival.c, channels/iax2-provision.c, apps/app_alarmreceiver.c, res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka) 2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming * apps/app_queue.c: fix buggy (and overly complex) loop used during reload of app_queue for static member list updating 2006-09-28 17:34 +0000 [r43918] Paul Cadach * channels/h323/ast_h323.cxx: Extend call establishment timeout 2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp * channels/chan_iax2.c: Make sure the pvt exists before accessing it again as it may have gone away (issue #7562 reported by Seb7 and issue #7939 reported by sorg) * main/cli.c: Warning be gone! 2006-09-28 16:41 +0000 [r43899] BJ Weschke * apps/app_queue.c: app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach) 2006-09-28 16:17 +0000 [r43893] Joshua Colp * apps/app_meetme.c: Stop the stream after waitstream returns so that our formats get restored. (issue #7370 reported by kryptolus) 2006-09-28 15:56 +0000 [r43877] Paul Cadach * channels/h323/ast_h323.cxx: Fix compiler warning 2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - tim_ringenbach reported and patched) * apps/app_queue.c: Autopause not working for queue members. (#8042 - jmls reported and patch) 2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force remote side to start media on outgoing PROGRESS message * include/asterisk/compiler.h: Put attribute tag at correct place 2006-09-28 11:03 +0000 [r43852] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c: fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed. 2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach * channels/h323/ast_h323.cxx: Do not open transmit channel until TCS is received * main/file.c: Don't warn on HOLD/UNHOLD control frames * main/file.c: Don't treat unknown control frames as voice 2006-09-27 20:21 +0000 [r43816] Tilghman Lesher * apps/app_voicemail.c: Avoid inability to lock directory log message by creating the directory ahead of time. (Issue 7631) 2006-09-27 19:44 +0000 [r43801-43803] Jason Parker * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS not being set under certain circumstances. Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string. Fix Background() to return -1 like Playback(), if no args are specified. 2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp * main/rtp.c: Compensate for out of order packets better if RFC2833 compensation is turned on. * channels/chan_iax2.c: Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right. 2006-09-27 16:55 +0000 [r43779] Russell Bryant * main/channel.c,res/res_features.c: Fix a problem that occurred if a user entered a digit that matched a bridge feature that was configured using multiple digits, and the digit that was pressed timed out in the feature digit timeout period. For example, if blind transfer is configured as '##', and a user presses just '#'. In this situation, the call would lock up and no longer pass any frames. (issue #7977 reported by festr, and issue #7982 reported by michaels and valuable input provided by mneuhauser and kuj. Fixed by me, with testing help and peer review from Joshua Colp). There are a couple of issues involved in this fix: 1) When ast_generic_bridge determines that there has been a timeout, it returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls ast_generic_bridge over again with the same timestamp for the next event. This results in an endless loop of nothing until the call is terminated. This is resolved by simply changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a timeout. 2) I also changed ast_channel_bridge such that if in the process of calculating the time until the next event, it knows a timeout has already occured, to immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the channels anyway. 3) In the process of testing the previous two changes, I ran into a problem in res_features where ast_channel_bridge would return because it determined that there was a timeout. However, ast_bridge_call in res_features would then determine by its own calculation that there was still 1 ms before the timeout really occurs. It would then proceed, and since the bridge broke out and did *not* return a frame, it interpreted this as the call was over and hung up the channels. The reason for this was because ast_bridge_call in res_features and ast_channel_bridge in channel.c were using different times for their calculations. channel.c uses the start_time on the bridge config, which is the time that the feature digit was recieved. However, res_features had another time, 'start', which was set right before calling ast_channel_bridge. 'start' will always be slightly after start_time in the bridge config, and sometimes enough to round up to one ms. This is fixed by making ast_bridge_call use the same time as ast_channel_bridge for the timeout calculation. ........ 2006-09-27 16:24 +0000 [r43775] Christian Richter * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn versioning, since Asterisk has it's own 2006-09-27 16:23 +0000 [r43774] Joshua Colp * channels/chan_sip.c: Make rfc2833compensate a global option. 2006-09-27 04:35 +0000 [r43756] Russell Bryant * apps/app_voicemail.c: Backport revision 43754 from the trunk, which removes an unused buffer from mm_login to close bug 8038, as well as addresses some formatting and coding guidelines issues in passing. Originally, I did not commit this to 1.4 since it is not necessarily fixing a bug. However, since the IMAP storage code is brand new, I decided it would be better to make the change here as well, in case someone has to work on this code to address issues in the very near future. I don't want to make unnecessary merge problems going to the trunk. 2006-09-27 02:32 +0000 [r43739] Steve Murphy * configs/extensions.ael.sample: This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse. 2006-09-26 20:56 +0000 [r43710] Russell Bryant * main/asterisk.c: Back in revision 4798, this message was changed from using ast_cli() to directly calling write(). During this change, checking if this was a remote console was removed. This caused this message about using "exit" or "quit" to exit an Asterisk console to come up in times where it did not make sense. This change restores the check to see if this is a remote console before printing the message. (fixes BE-65) 2006-09-26 20:47 +0000 [r43707] Joshua Colp * .cleancount, main/cli.c, channels/chan_sip.c, include/asterisk/channel.h: Use proper type to represent the group variable (issue #8025 reported by makoto) 2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant * channels/chan_sip.c: Add missing newline character in the warning message about deprecated TOS values in configuration. * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain mailbox definitions, don't introduce a length limit on the definition by using a 256 byte temporary storage buffer. Instead, make the temporary buffer just as big as it needs to be to hold the entire mailbox definition. (fixes BE-68) 2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp * channels/chan_local.c: Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo) * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980) 2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming * sounds/Makefile: update to use 1.4.3 core sounds, with corrected beep/beeperr/tt-monkeys files 2006-09-26 18:08 +0000 [r43650-43674] Jason Parker * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by Dan Austin. Maximum values were incorrect, which is why this is being put in 1.4 * channels/chan_skinny.c: Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a. This is technically a "new feature", but there are justifications for it. I found a bug with the recent rtp packetization changes, which caused the media setup to fail under certain circumstances, particularly when using allow=all, or having no allow= statements (globally or on the device). I could have either removed the rtp packetization features, or I could add proper codec support (which, without, I think most people would consider to be a bug anyways). 2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher * apps/app_voicemail.c: Should have moved these lines up in the merge, instead of removing them * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) delete=yes was ignored 2) maxmessages was ignored 2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, channels/h323/cisco-h225.asn: Fix ASN1 description of non-standard Cisco extensions * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport changes of trunk: 1) r43540: Avoid possible deadlock on channel destruction 2) r43590: Disable fastStart if requested by remote side 2006-09-25 15:23 +0000 [r43616] Jason Parker * sounds/Makefile: One more fix for sounds installation - this time for portability. Reported to asterisk-dev mailing list. 2006-09-25 14:52 +0000 [r43605] Steve Murphy * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from crashing if trying to play an OGG moh file. 2006-09-25 06:15 +0000 [r43582] Paul Cadach * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, channels/chan_h323.c: Merged revisions 43472,43495 from trunk 2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant * channels/iax2-provision.c: Fix a CLI command registration issue where an erroneous message claiming that "iax2 show provisioning" was already registered. This was because this command was registering itself as both the command, as well as the command it is deprecating. (issue #8022, reported by bjweeks, fixed by myself) * channels/chan_iax2.c:Check to see if the channel that is activating the IAXPEER function is actually an IAX2 channel before proceeding to process it to avoid crashing. (issue #8017, reported by admott, fixed by myself) 2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming * Makefile: don't output the 'build complete' message when the target being run is already going to do an installation 2006-09-22 22:12 +0000 [r43518] Jason Parker * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded properly. Remove reload support, since it doesn't actually...work. 2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy * pbx/pbx_ael.c: This commits a change to return MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all goes well for bug 8004 * pbx/pbx_ael.c: If the extensions.ael file not found, or unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. 2006-09-22 17:25 +0000 [r43492] Jason Parker * main/cli.c: Make sure we explicitly set the CLI command to not be deprecated, if it isn't. 2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming * sounds/Makefile: use rebuilt extra sounds * main/channel.c: all the Linux systems I have don't use '__m_count' for this field, so I don't know where this came from... 2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant * include/asterisk/threadstorage.h: backport the compatability fix to use attribute_malloc instaed of __attribute__ ((malloc)) * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN could not be configured (issue #8006, Mithraen) * main/frame.c: Suppress a compiler warning about the use of a potentially uninitialized variable. It couldn't actually happen, though. 2006-09-22 03:01 +0000 [r43469] Jason Parker * channels/chan_skinny.c: First shot at unload_module in chan_skinny.. More to come. 2006-09-21 23:50 +0000 [r43466] Matt O'Gorman * include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: updates for better compontent support 2006-09-21 23:24 +0000 [r43464] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we actually documented how the new features in res_odbc actually work. (Oops) 2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp * channels/chan_oss.c: Some more clean up in the load function for chan_oss (issue #8002 reported by Mithraen with minor mods by moi) * channels/chan_mgcp.c: Clean up chan_mgcp's module load function (issue #8001 reported by Mithraen with mods by moi) 2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming * main/Makefile, build_tools/strip_nonapi (added): add another attempt to strip non-API symbols from the final binary... script will need to be extended to work on non-Linux systems 2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher * apps/app_url.c: Fix documentation to reflect how Url() really works * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates 2006-09-21 Kevin P. Fleming * Asterisk 1.4.0-beta2 released. 2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming * main/Makefile: remove this change... it requires binutils 2.17 2006-09-20 23:19 +0000 [r43396] Jason Parker * build_tools/make_version: fix minor typo in the way version is handled 2006-09-20 Kevin P. Fleming * Asterisk 1.4.0-beta1 released.