2008-04-01 Russell Bryant * Asterisk 1.4.19 released. 2008-03-28 Russell Bryant * Asterisk 1.4.19-rc4 released. 2008-03-28 16:19 +0000 [r111658] Jason Parker * formats/format_wav_gsm.c: The file size of WAV49 does not need to be an even number. (closes issue #12128) Reported by: mdu113 Patches: 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: qwell, mdu113 2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher * doc/valgrind.txt: Update debugging text, since Valgrind eliminated the --log-file-exactly option. (Closes issue #12320) * main/acl.c: For FreeBSD, at least, the ifa_addr element could be NULL. (closes issue #12300) Reported by: festr Patches: acl.c.patch uploaded by festr (license 443) 2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy * apps/app_playback.c, main/pbx.c: These small documentation updates made in response to a query in asterisk-users, where a user was using Playback, but needed the features of Background, and had no idea that Background existed, or that it might provide the features he needed. I thought the best way to avert these kinds of queries was to provide "See Also" references in all three of "Background", "Playback", "WaitExten". Perhaps a project to do this with all related apps is in order. * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue #12302) Reported by: pj Tested by: murf These changes will set a channel variable ~~EXTEN~~ just before generating code for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, and ever after that, till the end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). The reason for this, is that because switches are coded using separate extensions to provide pattern matching, and jumping to/from these switch extensions messes up the ${EXTEN} value, which blows the minds of users. 2008-03-27 00:25 +0000 [r111245-111280] Jason Parker * main/frame.c: Put this flag back so we don't change the API. * main/frame.c: Remove excessive smoother optimization that was causing audio glitches (small "pops") after (about 200ms later) an "incorrectly" sized frame was received. While it would be very nice to keep this as optimized as possible, it makes no sense for the smoother to be dropping random bits of audio like this. Isn't that the whole point of a smoother? Closes issue #12093. 2008-03-26 19:55 +0000 [r111129] Joshua Colp * contrib/scripts/autosupport: Update autosupport script. (closes issue #12310) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) 2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines update UPGRADE notes to document usage of the script ........ 2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson * apps/app_voicemail.c: This code change is made just for clarification. It does exactly the same thing as before. It just doesn't look as wrong. * apps/app_voicemail.c: Add a lock to the vm_state structure and use the lock around mail_open calls to prevent concurrent access of the same mailstream. This, along with trunk's ability to configure TCP timeouts for IMAP storage will help to prevent crashes and hangs when using voicemail with IMAP storage. (closes issue #10487) Reported by: ewilhelmsen 2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): Merged revisions 111019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines add a script to make getting the iLBC source code simple for end users ........ 2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp * channels/chan_sip.c: If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall * channels/chan_iax2.c: Make sure that full video frames are sent whenever the 15 bit timestamp rolls over. (closes issue #11923) Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by mihai (license 94) 2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming * UPGRADE.txt: add note that the user will need to enable codec_ilbc to get it to build * codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h (removed), codecs/ilbc/getCBvec.c (removed), codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h (removed), codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed), codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c (removed), codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c (removed), codecs/ilbc/createCB.h (removed), CHANGES, codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed), codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c (removed), codecs/ilbc/iCBSearch.h (removed), codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c (removed), codecs/ilbc/hpOutput.h (removed), codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h (removed), codecs/ilbc/iCBConstruct.c (removed), codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h (removed), codecs/ilbc/syntFilter.h (removed), codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c (removed): Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ 2008-03-25 22:51 +0000 [r110779] Jason Parker * cdr/cdr_custom.c: Make file access in cdr_custom similar to cdr_csv. Fixes issue #12268. Patch borrowed from r82344 2008-03-25 20:03 +0000 [r110727] Jeff Peeler * channels/chan_sip.c: This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. 2008-03-25 15:40 +0000 [r110635] Mark Michelson * channels/chan_sip.c: When reverting a commit, I accidentally left in this bit which was an experiment to see what would happen. It passed the compile test, and I didn't notice I had left this change in too. So this is a revert of a revert...sort of. 2008-03-25 14:37 +0000 [r110628] Joshua Colp * include/asterisk/options.h, main/asterisk.c, Makefile, main/app.c: Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet 2008-03-24 19:17 +0000 [r110618] Mark Michelson * channels/chan_sip.c: This is a revert for revision 108288. The reason is that that revision was not for an actual bug fix per se, and so it really should not have been in 1.4 in the first place. Plus, people who compile with DO_CRASH are more likely to encounter a crash due to this change. While I think the usage of DO_CRASH in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4 and should be done instead in a developer branch based on trunk so that all scheduler functions are fixed at once. I also am reverting the change to trunk and 1.6 since they also suffer from the DO_CRASH potential. (closes issue #12272) Reported by: qq12345 2008-03-24 17:34 +0000 [r110614] Russell Bryant * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message. 2008-03-21 14:32 +0000 [r110474] Jason Parker * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on mips platforms either. (closes issue #12270) Reported by: zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) 2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread. This really should not make a difference except in very rare cases. That case would be that all of the channels in autoservice are not generating any frames. In that case, this change reduces the potential amount of time that a thread waits in ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning of its loop. (closes issue #12266, reported by dimas) * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 110335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines Fix some very broken code that was introduced in 1.2.26 as a part of the security fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address structure that a background thread continuously updates. However, in these cases, a stack variable was passed. That means that the dnsmgr thread would be continuously writing to bogus memory. ........ * apps/app_meetme.c: Fix a bug where when calls on the trunk side hang up while on hold, the state is not properly reflected. (closes issue #11990, reported by anakaoka, patched by me) 2008-03-19 20:33 +0000 [r110083] Mark Michelson * apps/app_chanspy.c: Add a missing unlock in the case that memory allocation fails in app_chanspy. Thanks to Russell for confirming that this was an issue. 2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp * res/res_musiconhold.c: Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior. (closes issue #11663) Reported by: junky * main/rtp.c: Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) 2008-03-19 17:12 +0000 [r109973] Jason Parker * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml (added): People report bugs about Asterisk crashing with DO_CRASH enabled was getting a little silly... Now we only show certain cflags when you run configure with --enable-dev-mode (corresponding menuselect change to follow) 2008-03-19 15:41 +0000 [r109908] Steve Murphy * main/config.c: (closes issue #11442) Reported by: tzafrir Patches: 11442.patch uploaded by murf (license 17) Tested by: murf I didn't give tzafrir very much time to test this, but if he does still have remaining issues, he is welcome to re-open this bug, and we'll do what is called for. I reproduced the problem, and tested the fix, so I hope I am not jumping by just going ahead and committing the fix. The problem was with what file_save does with templates; firstly, it tended to print out multiple options: [my_category](!)(templateref) instead of [my_category](!,templateref) which is fixed by this patch. Nextly, the code to suppress output of duplicate declarations that would occur because the reader copies inherited declarations down the hierarchy, was not working. Thus: [master-template](!) mastervar = bar [template](!,master-template) tvar = value [cat](template) catvar = val would be rewritten as: ;! ;! Automatically generated configuration file ;! Filename: experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! [master-template](!) mastervar = bar [template](!,master-template) mastervar = bar tvar = value [cat](template) mastervar = bar tvar = value catvar = val This has been fixed. Since the config reader 'explodes' inherited vars into the category, users may, in certain circumstances, see output different from what they originally entered, but it should be both correct and equivalent. 2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant * main/utils.c: Tweak spacing in a recent change because I'm very picky. * apps/app_chanspy.c: Fix one place where the chanspy datastore isn't removed from a channel. (issue #12243, reported by atis, patch by me) 2008-03-18 20:52 +0000 [r109713] Mark Michelson * apps/app_queue.c: This patch makes it so that all queue member status changes are handled through device state code. This removes several problems people were seeing where their queue members would get into an "unknown" state. Huge props go to atis on this one since he was the one who found the code section that was causing the problem and proposed the solution. I just wrote what he suggested :) (closes issue #12127) Reported by: atis Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested by: atis, jvandal 2008-03-18 19:23 +0000 [r109648] Jason Parker * codecs/log2comp.h: Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations. (closes issue #12253) Reported by: fossil Patches: log2comp.patch uploaded by fossil (license 140) 2008-03-18 17:58 +0000 [r109575] Mark Michelson * channels/chan_agent.c: Make sure an agent doesn't try to send dtmf to a NULL channel closes issue #12242 Reported by Yourname 2008-03-18 Russell Bryant * Asterisk 1.4.19-rc3 released. 2008-03-18 16:25 +0000 [r109482] Terry Wilson * include/asterisk/astobj.h: Fix character string being treated ad format string 2008-03-18 15:10 +0000 [r109393] Jason Parker * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) | 3 lines Do not return with a successful authentication if the From header ends up empty. (AST-2008-003) ........ 2008-03-18 14:58 +0000 [r109386] Joshua Colp * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) 2008-03-18 06:37 +0000 [r109309] Steve Murphy * pbx/ael/ael-test/ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t1/a.ael (added), pbx/ael/ael-test/ael-ntest23/t1/b.ael (added), pbx/ael/ael-test/ael-ntest23/t1/c.ael (added), pbx/ael/ael-test/ael-ntest23/t2/d.ael (added), pbx/ael/ael-test/ael-ntest23/t2/e.ael (added), pbx/ael/ael-test/ael-ntest23/t2/f.ael (added), pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c, pbx/ael/ael-test/ael-ntest23/t3/g.ael (added), pbx/ael/ael-test/ael-ntest23/t3/h.ael (added), pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex, pbx/ael/ael-test/ael-ntest23/t3/j.ael (added), pbx/ael/ael-test/ael-ntest23/qq.ael (added), pbx/ael/ael-test/ael-ntest23/t1 (added), pbx/ael/ael-test/ael-ntest23/t2 (added), pbx/ael/ael-test/ael-ntest23/t3 (added), pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes issue #11903) Reported by: atis Many thanks to atis for spotting this problem and reporting it. The fix was to straighten out how items are placed on and removed from the file stack. Regressions as well as the provided test case helped to straighten out all code paths. valgrind was used to make sure all memory allocated was freed. Sorry for not solving this earlier. I got distracted. Added the ntest23 regression test, which is mainly a copy of ntest22, but with a few juicy errors thrown in, to replicate the kind of error that atis spotted. 2008-03-17 22:05 +0000 [r109226] Mark Michelson * main/utils.c: Fix a logic flaw in the code that stores lock info which is displayed via the "core show locks" command. The idea behind this section of code was to remove the previous lock from the list if it was a trylock that had failed. Unfortunately, instead of checking the status of the previous lock, we were referencing the index immediately following the previous lock in the lock_info->locks array. The result of this problem, under the right circumstances, was that the lock which we currently in the process of attempting to acquire could "overwrite" the previous lock which was acquired. While this does not in any way affect typical operation, it *could* lead to misleading "core show locks" output. 2008-03-17 17:55 +0000 [r109171] Michiel van Baak * channels/chan_skinny.c: Update the directory of placed calls on skinny phones when dialing a channel that does not provide progress (analog ZAP lines) The phone does handle the double update on calls to channels that do provide progress and wont insert duplicate items (closes issue #12239) Reported by: DEA Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) 2008-03-17 16:24 +0000 [r109107] Joshua Colp * channels/chan_sip.c: 200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down. (closes issue #12208) Reported by: atrash 2008-03-17 15:15 +0000 [r109057] Jason Parker * main/file.c: Backport revision 106439 from trunk. I didn't realize this was broken in 1.4 as well. Closes issue #12222. 2008-03-17 14:18 +0000 [r109012] Mark Michelson * apps/app_chanspy.c: Make sure that we release the lock on the spyee channel if the spyee or spy has hung up (closes issue #12232) Reported by: atis 2008-03-16 21:47 +0000 [r108961] Michiel van Baak * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE (closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) 2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant * channels/chan_oss.c: Fix a channel name issue. chan_oss registers the "Console" channel type, but it created channels with an "OSS" prefix. (closes issue #12194, reported by davidw, patched by me) * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to start networking before asterisk, as well. (closes issue #12200, reported by and change suggested by reinerotto) 2008-03-14 16:44 +0000 [r108737] Mark Michelson * channels/chan_sip.c: Fix a race condition in the SIP packet scheduler which could cause a crash. chan_sip uses the scheduler API in order to schedule retransmission of reliable packets (such as INVITES). If a retransmission of a packet is occurring, then the packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if a response is received from the packet as previously transmitted, then when we ACK the response, we will remove the packet from the scheduler and free the packet. The problem is that both the ACK function and retrans_pkt attempt to acquire the same lock at the beginning of the function call. This means that if the ACK function acquires the lock first, then it will free the packet which retrans_pkt is about to read from and write to. The result is a crash. The solution: 1. If the ACK function fails to remove the packet from the scheduler and the retransmit id of the packet is not -1 (meaning that we have not reached the maximum number of retransmissions) then release the lock and yield so that retrans_pkt may acquire the lock and operate. 2. Make absolutely certain that the ACK function does not recursively lock the lock in question. If it does, then releasing the lock will do no good, since retrans_pkt will still be unable to acquire the lock. (closes issue #12098) Reported by: wegbert (closes issue #12089) Reported by: PTorres Patches: 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested by: jvandal 2008-03-14 14:29 +0000 [r108682] Jason Parker * res/res_musiconhold.c: Fix a potential segfault if chan (or chan->music_state) is NULL. Closes issue #12210, credit to edantie for pointing this out. 2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h: Fix another issue that was causing crashes in chanspy. This introduces a new datastore callback, called chan_fixup(). The concept is exactly like the fixup callback that is used in the channel technology interface. This callback gets called when the owning channel changes due to a masquerade. Before this was introduced, if a masquerade happened on a channel being spyed on, the channel pointer in the datastore became invalid. (closes issue #12187) (reported by, and lots of testing from atis) (props to file for the help with ideas) * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom phones to blink when an extension that has been subscribed to goes on hold. Otherwise, they just stay on like it does when an extension is in use. (closes issue #11263) Reported by: russell Patches: notify_hold.rev1.txt uploaded by russell (license 2) Tested by: russell * apps/app_followme.c: Fix a couple uses of sprintf. The second one could actually cause an overflow of a stack buffer. It's not a security issue though, it only depends on your configuration. 2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip. The scheduler callback will always return 0. This means that this id is never rescheduled, so it makes no sense to loop trying to delete the id from the scheduler queue. If we fail to remove the item from the queue once, it will fail every single time. (Yes I realize that in this case, the macro would exit early because the id is set to -1 in the callback, but it still makes no sense to use that macro in favor of calling ast_sched_del once and being done with it) This is the first of potentially several such fixes. * include/asterisk/sched.h: Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when it is appropriate and when it is not appropriate to use it. I also removed the part of the debug message that mentions that this is probably a bug because there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g. when the scheduler callback manually reschedules with a new id instead of returning non-zero to tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro is probably being used in several places where it shouldn't be. Also removed the redundant line, file, and function information since that is provided by ast_log. 2008-03-12 19:57 +0000 [r108135] Russell Bryant * apps/app_chanspy.c, main/channel.c: (closes issue #12187, reported by atis, fixed by me after some brainstorming on the issue with mmichelson) - Update copyright info on app_chanspy. - Fix a race condition that caused app_chanspy to crash. The issue was that the chanspy datastore magic that was used to ensure that spyee channels did not disappear out from under the code did not completely solve the problem. It was actually possible for chanspy to acquire a channel reference out of its datastore to a channel that was in the middle of being destroyed. That was because datastore destruction in ast_channel_free() was done near the end. So, this left the code in app_chanspy accessing a channel that was partially, or completely invalid because it was in the process of being free'd by another thread. The following sort of shows the code path where the race occurred: ============================================================================= Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) --------------------------------------||------------------------------------- ast_channel_free() || - remove channel from channel list || - lock/unlock the channel to ensure || that no references retrieved from || the channel list exist. || --------------------------------------||------------------------------------- || channel_spy() - destroy some channel data || - Lock chanspy datastore || - Retrieve reference to channel || - lock channel || - Unlock chanspy datastore --------------------------------------||------------------------------------- - destroy channel datastores || - call chanspy datastore d'tor || which NULL's out the ds' || - Operate on the channel ... reference to the channel || || - free the channel || || || - unlock the channel --------------------------------------||------------------------------------- ============================================================================= 2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming * channels/chan_sip.c: if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP closes issue #11475 Reported by: andrebarbosa 2008-03-12 18:26 +0000 [r108083] Joshua Colp * apps/app_mixmonitor.c, include/asterisk/audiohook.h, main/audiohook.c: Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox 2008-03-12 16:59 +0000 [r108031] Russell Bryant * main/channel.c: Destroy the channel lock after the channel datastores. (inspired by issue #12187) 2008-03-12 01:52 +0000 [r107877] Tilghman Lesher * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql: Document all of the possible realtime fields 2008-03-11 23:37 +0000 [r107714-107826] Jason Parker * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql ODBC voicemail. (closes issue #12186) Reported by: jsmith Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license 15) * channels/chan_gtalk.c: Copy voicemail dependency logic for res_adsi to chan_gtalk (for jabber). (closes issue #12014) Reported by: junky 2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming * Makefile.rules, channels/Makefile: get chan_vpb to build properly in dev mode 2008-03-11 20:47 +0000 [r107712] Jason Parker * apps/app_voicemail.c: Add a newline on a log 2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp * res/res_features.c: Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted. (closes issue #11707) Reported by: iam * apps/app_meetme.c: Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened. (closes issue #12136) Reported by: aragon * apps/app_exec.c: Fix a minor spelling error. (closes issue #12183) Reported by: darrylc 2008-03-11 Russell Bryant * Asterisk 1.4.19-rc2 released. 2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming * apps/app_rpt.c: backport a fix from trunk * channels/misdn/isdn_lib.c, codecs/Makefile, channels/chan_misdn.c: fix various other problems found by gcc 4.3 * configure, include/asterisk/autoconfig.h.in, configure.ac, apps/app_sms.c: stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3 * configure, main/Makefile, configure.ac, makeopts.in: check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview * configure, configure.ac: fix small bug in IMAP toolkit testing * main/udptl.c, utils/Makefile, main/Makefile, main/editline/readline.c, pbx/Makefile: fix up various compiler warnings found with gcc-4.3: - the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function) - main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement - main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur - main/editline/readline.c had an unused variable 2008-03-11 00:59 +0000 [r107290] Terry Wilson * channels/chan_sip.c: If we fail to alloc a channel, we should re-lock the pvt structure before returning. 2008-03-10 21:32 +0000 [r107230] Tilghman Lesher * main/pbx.c: Use non-global storage for eswitch 2008-03-10 20:27 +0000 [r107173] Jason Parker * channels/chan_zap.c: Make sure to reenable echo can after a "failed" (canceled, etc) three-way call. (closes issue #11335) Reported by: rebuild 2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant * main/pbx.c: Fix another bug specifically related to asynchronous call origination. Once the PBX is started on the channel using ast_pbx_start(), then the ownership of the channel has been passed on to another thread. We can no longer access it in this code. If the channel gets hung up very quickly, it is possible that we could access a channel that has been free'd. (inspired by BE-386) * main/pbx.c: Fix some bugs related to originating calls. If the code failed to start a PBX on the channel (such as if you set a call limit based on the system's load average), then there were cases where a channel that has already been free'd using ast_hangup() got accessed. This caused weird memory corruption and crashes to occur. (fixes issue BE-386) (much debugging credit goes to twilson, final patch written by me) * main/channel.c: Resolve a compiler warning. * main/channel.c: Fix a race condition where the generator can go away (closes issue #12175, reported by edantie, patched by me) 2008-03-10 14:33 +0000 [r107016] Joshua Colp * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader 2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming * channels/chan_zap.c: don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down" 2008-03-07 22:51 +0000 [r106895] Russell Bryant * apps/app_meetme.c: Only start the SLA thread if SLA has actually been configured. 2008-03-07 22:14 +0000 [r106842] Jason Parker * main/editline/Makefile.in: Fix hardcoded grep in editline, were GNU grep is required. (closes issue #12124) Reported by: dmartin 2008-03-07 19:32 +0000 [r106788] Joshua Colp * main/channel.c: Ignore source update control frame. (closes issue #12168) Reported by: plack 2008-03-07 17:16 +0000 [r106704] Russell Bryant * include/asterisk/sched.h: Change a warning message to a debug message. This is happening quite frequently, and it is not worth spamming users with these messages unless we are pretty confident that it should never happen. As it stands today, it _will_ and _does_ happen and until that gets cleaned up a reasonable amount on the development side, let's not spam the logs of everyone else. (closes issue #12154) 2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher * apps/app_voicemail.c: Warn the user when a temporary greeting exists (Closes issue #11409) * main/rtp.c: Properly initialize rtp->schedid (Closes issue #12154) * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c, apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c, funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, main/manager.c: Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) 2008-03-06 22:10 +0000 [r106437] Mark Michelson * main/pbx.c: Quell an annoying message that is likely to print every single time that ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial allocates the cdr for the channel, so it should be expected that the channel will have a cdr on it. Thanks to joetester on IRC for pointing this out 2008-03-06 04:40 +0000 [r106328] Tilghman Lesher * sounds/Makefile: Upgrade to the next release of sounds 2008-03-05 22:37 +0000 [r106237] Russell Bryant * channels/chan_iax2.c: Fix a potential deadlock and a few different potential crashes. (closes issue #12145, reported by thiagarcia, patched by me) 2008-03-05 22:32 +0000 [r106235] Joshua Colp * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c, apps/app_dial.c, main/channel.c, channels/chan_phone.c, main/dial.c, channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, main/file.c, channels/chan_alsa.c, apps/app_followme.c, include/asterisk/frame.h: Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas 2008-03-05 21:12 +0000 [r106178] Michiel van Baak * doc/realtime.txt: document var_metric so no bugreports will come in when it's actually a configuration issue. (issue #12151) Reported and patched by: caio1982 1.4 patch by me 2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming * channels/chan_zap.c: when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one (closes issue #11917) Reported by: mavetju Tested by: mavetju 2008-03-05 15:17 +0000 [r106015] Tilghman Lesher * channels/chan_sip.c, include/asterisk/sched.h: Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log. (closes issue #12140) Reported by: slavon Patches: sch2.patch uploaded by slavon (license 288) (Patch slightly modified by me) 2008-03-05 01:52 +0000 [r105932] Russell Bryant * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. 2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp * main/rtp.c: In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source. * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson 2008-03-04 Russell Bryant * Asterisk 1.4.19-rc1 released. 2008-03-04 04:31 +0000 [r105591] Russell Bryant * main/pbx.c: Backport a minor bug fix from trunk that I found while doing random code cleanup. Properly break out of the loop when a context isn't found when verify that includes are valid. 2008-03-03 18:06 +0000 [r105572] Jason Parker * res/snmp/agent.c: Fix type for astNumChannels. (closes issue #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg (license 192) 2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant * channels/chan_local.c: In the case of an ast_channel allocation failure, take the local_pvt out of the pvt list before destroying it. * channels/chan_local.c: Fix a potential memory leak of the local_pvt struct when ast_channel allocation fails. Also, in passing, centralize the code necessary to destroy a local_pvt. * main/autoservice.c: Update the copyright information for autoservice. Most of the code in this file now is stuff that I have written recently ... * main/asterisk.c, main/channel.c, include/asterisk.h, main/autoservice.c: Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) 2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp * main/channel.c: It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg Patches: v1-11911.patch uploaded by dimas (license 88) Tested by: tbsky * channels/chan_sip.c: Add a comment to describe some logic. (closes issue #12120) Reported by: flefoll Patches: chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244) 2008-02-29 23:34 +0000 [r105409] Russell Bryant * main/autoservice.c: Fix a major bug in autoservice. There was a race condition in the handling of the list of channels in autoservice. The problem was that it was possible for a channel to get removed from autoservice and destroyed, while the autoservice thread was still messing with the channel. This led to memory corruption, and caused crashes. This explains multiple backtraces I have seen that have references to autoservice, but do to the nature of the issue (memory corruption), could cause crashes in a number of areas. (fixes the crash in BE-386) (closes issue #11694) (closes issue #11940) The following issues could be related. If you are the reporter of one of these, please update to include this fix and try again. (potentially fixes issue #11189) (potentially fixes issue #12107) (potentially fixes issue #11573) (potentially fixes issue #12008) (potentially fixes issue #11189) (potentially fixes issue #11993) (potentially fixes issue #11791) 2008-02-29 14:47 +0000 [r105326] Philippe Sultan * res/res_jabber.c: Fix a potential memory leak 2008-02-29 14:34 +0000 [r105296] Tilghman Lesher * apps/app_voicemail.c: If the message file does not exist, just return harmlessly, instead of crashing. (Closes issue #12108) 2008-02-29 13:48 +0000 [r105261] Joshua Colp * apps/app_voicemail.c: Bump up the size of the uniqueid variable. (closes issue #12107) Reported by: asgaroth 2008-02-29 13:05 +0000 [r105209] Philippe Sultan * res/res_jabber.c: Automatically create new buddy upon reception of a presence stanza of type subscribed. (closes issue #12066) Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by phsultan (license 73) trunk-12066-1.diff uploaded by phsultan (license 73) Tested by: ffadaie, phsultan 2008-02-28 22:23 +0000 [r105116] Russell Bryant * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock tracking code that was discovered by mmichelson. The issue is that if the lock history array was full, then the functions to mark a lock as acquired or not would adjust the stats for whatever lock is at the end of the array, which may not be itself. So, do a sanity check to make sure that we're updating lock info for the proper lock. (This explains the bizarre stats on lock #63 in BE-396, thanks Mark!) 2008-02-28 21:56 +0000 [r105113] Tilghman Lesher * contrib/init.d/rc.debian.asterisk: Update init script for LSB compat (closes issue #9843) Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by: paravoid 2008-02-28 20:11 +0000 [r105059] Mark Michelson * apps/app_queue.c: When using autofill, members who are in use should be counted towards the number of available members to call if ringinuse is set to yes. Thanks to jmls who brought this issue up on IRC 2008-02-28 19:20 +0000 [r104920-105005] Jason Parker * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into applications, if we get NULL. This protects against possible segfaults in applications that may try to use data before checking length (ast_strdupa'ing it, for example) (closes issue #12100) Reported by: foxfire Patches: 12100-nullappargs.diff uploaded by qwell (license 4) * channels/chan_skinny.c: According to a video at www.cisco.com, the 7921G supports 6 line appearances. 2008-02-28 00:05 +0000 [r104868] Tilghman Lesher * main/Makefile, build_tools/strip_nonapi: Compatibility fix for PPC64 (closes issue #12081) Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412) Tested by: jcollie, Corydon76 2008-02-27 21:49 +0000 [r104841] Mark Michelson * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a lockable list. This is because in some cases, the ast_dial may exist in multiple threads due to asynchronous execution of its application, and I found some cases where race conditions could exist. 2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been cleared yet. (closes issue #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by putnopvut (license 60) Tested by: jvandal 2008-02-27 20:56 +0000 [r104787] Joshua Colp * apps/app_chanspy.c: Don't loop around infinitely trying to spy on our own channel, and don't forget to free/detach the datastore upon hangup of the spy. 2008-02-27 20:36 +0000 [r104783] Mark Michelson * main/file.c: Bump a couple of more buffers up by 2 so that annoying warnings aren't generated like crazy on every fileexists_core call. 2008-02-27 18:15 +0000 [r104704] Tilghman Lesher * main/manager.c: Ensure the session ID can't be 0. 2008-02-27 17:41 +0000 [r104665] Joshua Colp * main/file.c: Bump up the buffer by 2. 2008-02-27 17:33 +0000 [r104625] Russell Bryant * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get stuck in an infinite loop without being able to detect that the calling channel hung up. (closes issue #12076, reported by junky, patched by me) 2008-02-27 17:26 +0000 [r104598] Jason Parker * res/res_features.c: Inherit language from the transfering channel on a blind transfer. (closes issue #11682) Reported by: caio1982 Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, victoryure 2008-02-27 17:07 +0000 [r104596] Joshua Colp * main/loader.c: Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock. (closes issue #12080) Reported by: ChaseVenters 2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming * main/file.c: fallback to standard English prompts properly when using new prompt directory layout (closes issue #11831) Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines) 2008-02-27 16:45 +0000 [r104591] Russell Bryant * channels/chan_zap.c: When we receive a known alarm, make sure that the unknown alarm flag is not still set to make sure that when we come back out of alarm, it gets reported in the log and manager interface (after discussion with tzafrir on the -dev list) 2008-02-27 15:52 +0000 [r104536] Joshua Colp * res/res_smdi.c: Only stop the MWI monitor thread if it was actually started. (closes issue #12086) Reported by: francesco_r 2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant * apps/app_chanspy.c: Avoid some recursion in the cleanup code for the chanspy datastore (closes issue #12076, reported by junky, patched by me) * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as an alarm. However, Asterisk 1.4 does not know what to do with these alarms. Only Asterisk 1.6 cares about it. So, if we get an unknown alarm in chan_zap, don't generate confusing log messages about it. 2008-02-26 18:26 +0000 [r104132-104141] Jason Parker * Makefile: Add badshell to .PHONY target (thanks Kevin) * Makefile: Since all shells aren't as awesome as bash, we have to fail if somebody tries to use a literal "~" in DESTDIR. * sounds/Makefile: Revert previous abspath change. ...abspath is new in GNU make 3.81. I feel so...defeated. Must find new fix! * sounds/Makefile: Fix a very bizarre issue we were seeing with our buildbot when using a DESTDIR that wasn't an absolute path (such as DESTDIR=~/asterisk-1.4). Apparently what was happening, was that some of the targets were being expanded to the full path, so $@ ended up being /root/asterisk-1.4/[...]/ rather than ~/asterisk-1.4/[...]/ It appears that this may be a new "feature" in GNU make. (*cough* http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*) 2008-02-26 00:25 +0000 [r104119] Russell Bryant * include/asterisk/smdi.h, apps/app_voicemail.c, channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample: Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) 2008-02-26 00:03 +0000 [r104111] Jason Parker * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris. (closes issue #12050) Reported by: asgaroth Patches: 12050.patch uploaded by putnopvut (license 60) 2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant * apps/app_chanspy.c: This patch fixes some pretty significant problems with how app_chanspy handles pointers to channels that are being spied upon. It was very likely that a crash would occur if the channel being spied upon hung up. This was because the current ast_channel handling _requires_ that the object is locked or else it could disappear at any time (except in the owning channel thread). So, this patch uses some channel datastore magic on the spied upon channel to be able to detect if and when the channel goes away. (closes issue #11877) (patch written by me, but thanks to kpfleming for the idea, and to file for review) * main/utils.c: Improve the lock tracking code a bit so that a bunch of old locks that threads failed to lock don't sit around in the history. When a lock is first locked, this checks to see if the last lock in the list was one that was failed to be locked. If it is, then that was a lock that we're no longer sitting in a trylock loop trying to lock, so just remove it. (inspired by issue #11712) 2008-02-25 21:37 +0000 [r104095] Joshua Colp * channels/chan_sip.c: Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device. (closes issue #9044) Reported by: queuetue Patches: sip-gui-friend.diff uploaded by qwell (license 4) 2008-02-25 21:31 +0000 [r104094] Tilghman Lesher * apps/app_voicemail.c: If the destination folder is full, don't delete a message when exiting. (closes issue #12065) Reported by: selsky Patch by: (myself) 2008-02-25 20:49 +0000 [r104092] Jason Parker * main/config.c: Allow the use of #include and #exec in situations where the max include depth was only 1. Specifically, this fixes using #include and #exec in extconfig.conf. This was basically caused because the config file itself raises the include level to 1. I opted not to raise the include limit, because recursion here could cause very bizarre behavior. Pointed out, and tested by jmls (closes issue #12064) 2008-02-25 18:38 +0000 [r104086] Russell Bryant * channels/chan_agent.c: Ensure that the channel doesn't disappear in agent_logoff(). If it does, it could cause a crash. (fixes the crash reported in BE-396) 2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp * channels/chan_sip.c: If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog. (closes issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file (license 11) * channels/chan_sip.c: Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank. (closes issue #12061) Reported by: flefoll Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244) 2008-02-22 22:45 +0000 [r104037] Tilghman Lesher * channels/chan_sip.c: Backwards debug message. (closes issue #12052) Reported by: flefoll Patches: chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license 244) 2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson * channels/chan_zap.c: And as a followup to revision 104026, completely remove event-related calls from a section of code where we know there was no event to handle or get. * channels/chan_zap.c: Remove an incorrect debug message. It reported that it had received a specific event and tried to report which event was received. What actually was happening was that it was reporting the number of bytes returned from a call to read(). Thanks to Jared Smith for bringing the issue up on IRC 2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming * main/manager.c: reduce the likelihood that HTTP Manager session ids will consist of primarily '1' bits 2008-02-20 22:32 +0000 [r103956] Mark Michelson * apps/app_queue.c: Clear up confusion when viewing the QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the user's perspective, the queue does exist, we shouldn't tell them we couldn't find the queue. Instead since it is a dead queue, report a 0 waiting count This issue was brought up on IRC by jmls 2008-02-20 22:06 +0000 [r103953] Joshua Colp * channels/chan_zap.c: Don't wait for additional digits when overlap dialing is enabled if the setup message contains the sending_complete information element. (closes issue #11785) Reported by: klaus3000 Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by klaus3000 (license 65) 2008-02-20 21:40 +0000 [r103904] Mark Michelson * channels/chan_local.c: Fix a crash if the channel becomes NULL while attempting to lock it. (closes issue #12039) Reported by: danpwi 2008-02-20 17:53 +0000 [r103845] Tilghman Lesher * main/stdtime/localtime.c: Compat fix for Solaris (closes issue #12022) Reported by: asgaroth Patches: 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14) Tested by: asgaroth 2008-02-19 20:28 +0000 [r103823] Joshua Colp * channels/h323/ast_h323.cxx: Send CallerID Name in setup message. (closes issue #11241) Reported by: tusar Patches: h323id_as_callerid_name.patch uploaded by tusar (license 344) 2008-02-19 20:02 +0000 [r103821] Russell Bryant * channels/chan_local.c: Account for the fact that the "other" channel can disappear while the local pvt is not locked. (fixes a problem introduced in rev 100581) (closes issue #12012) Reported by: stevedavies Patch by me 2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp * configure, configure.ac: Don't look for launchd when cross compiling. (closes issue #12029) Reported by: ovi * channels/chan_sip.c: Fix building of chan_sip. 2008-02-19 10:27 +0000 [r103806] Olle Johansson * channels/chan_sip.c: Make sure we send error replies correctly by checking the via header. 2008-02-18 23:56 +0000 [r103801] Joshua Colp * main/channel.c: Ensure that emulated DTMFs do not get interrupted by another begin frame. (closes issue #11740) Reported by: gserra Patches: v1-11740.patch uploaded by dimas (license 88) (closes issue #11955) Reported by: tsearle (closes issue #10530) Reported by: xmarksthespot 2008-02-18 22:28 +0000 [r103790-103795] Jason Parker * channels/chan_zap.c: Fix previous commit so that we actually disable echocanbridged if echocancel is off. * channels/chan_zap.c: Correct a message when echocancelwhenbridged is on, but echocancel is not. Issue #12019 2008-02-18 20:52 +0000 [r103786] Mark Michelson * main/app.c: There was an invalid assumption when calculating the duration of a file that the filestream in question was created properly. Unfortunately this led to a segfault in the situation where an unknown format was specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream was written correctly or else assume a zero duration. (closes issue #12021) Reported by: jakep Tested by: putnopvut 2008-02-18 17:31 +0000 [r103780] Tilghman Lesher * main/rtp.c, channels/chan_sip.c: When a SIP channel is being auto-destroyed, it's possible for it to still be in bridge code. When that happens, we crash. Delay the RTP destruction until the bridge is ended. (closes issue #11960) Reported by: norman Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) Tested by: norman 2008-02-18 16:37 +0000 [r103770] Mark Michelson * channels/chan_zap.c: Fix a linked list corruption that under the right circumstances could lead to a looped list, meaning it will traverse forever. (closes issue #11818) Reported by: michael-fig Patches: 11818.patch uploaded by putnopvut (license 60) Tested by: michael-fig 2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp * main/asterisk.c: Backport fix from issue #9325. (closes issue #11980) Reported by: rbrunka * channels/chan_sip.c: Don't care if the extension given doesn't exist for subscription based MWI. 2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race condition (closes issue #11780) Reported by: guillecabeza Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license 380) * main/loader.c: In the case that you try to directly reload a module has returned AST_MODULE_LOAD_DECLINE, log a message indicating that the module is not fully initialized and must be initialized using "module load". * main/loader.c: Don't attempt to execute the reload callback for a module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash that was reported against chan_console in trunk. (closes issue #11953, reported by junky, fixed by me) 2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson * doc/imapstorage.txt, configure, configure.ac: Final round of changes for configure script logic for IMAP Now if a directory is specified, then we will search that directory for a source installation of the IMAP toolkit. If none is found, then we will use that directory as the basis for detecting a package installation of the IMAP c-client. If that check fails, then configure will fail. * configure, configure.ac: Fix a bit of wrong logic in the configure script that caused problems when trying to configure without IMAP. Patch suggestion from phsultan, but I modified it slightly. (closes issue #12003) Reported by: pj Tested by: putnopvut * doc/imapstorage.txt, configure, configure.ac: I apparently misunderstood one of the requirements of this configure change. Now, if a source directory is specified with the --with-imap option, and a valid source installation is not detected there, then configure will fail and will not check for a package installation. * doc/imapstorage.txt: Make a small clarification in the documentation * doc/imapstorage.txt: Update documentation regarding configuration of IMAP * apps/app_voicemail.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Change to the configure logic regarding IMAP. Prior to this commit, if you wished to configure Asterisk with IMAP support, you would use the --with-imap configure switch in one of the following two ways: --with-imap=/some/directory would look in the directory specified for a UW IMAP source installation --with-imap would assume that you had imap-2004g installed in .. relative to the Asterisk source With this set of changes the two above options still work the same, but there are two new behaviors, too. --with-imap=system will assume that you have -libc-client.so where you store your shared objects and will attempt to find c-client headers in your include path either in the imap or c-client directory. If either of the two original methods of specifying the imap option should fail, then the check for --with-imap =system will be performed in addition. It is only after this "system" check that failure can happen. * apps/app_voicemail.c: Fix build for non-IMAP builds * apps/app_voicemail.c: Fix the new message count if delete=yes when using IMAP storage. (closes issue #11406) Reported by: jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license 50) Tested by: jaroth 2008-02-14 19:51 +0000 [r103683-103684] Jason Parker * funcs/func_cdr.c: swap location for this.. * funcs/func_cdr.c: Document the 'l' option to the CDR() function. (Thanks voipgate for pointing out the option, and Leif for providing text for it.) Closes issue #11695. 2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher * channels/chan_agent.c: We aren't talking to ourselves; we're talking to someone else. (closes issue #11771) Reported by: msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, msetim * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is not loaded (which is a symbol dependency) (closes issue #11760) Reported by: non-poster Patches: 20080114__bug11760.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, non-poster, jamesgolovich 2008-02-12 22:24 +0000 [r103503-103504] Jason Parker * main/asterisk.c: revert accidental change from last commit. oops * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition that was impossible. 2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp * channels/chan_sip.c: Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones. (closes issue #11977) Reported by: pj * apps/app_meetme.c: If entering a conference with the 'w' option ensure that we can't listen or speak until the marked user appears. (closes issue #11835) Reported by: alanmcmillan 2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming * configs/zapata.conf.sample: improve 2BCT documentation a bit (thanks Jared) 2008-02-09 06:23 +0000 [r103197] Tilghman Lesher * apps/app_voicemail.c: Commit fix for being unable to send voicemail from VoiceMailMain Reported by: William F Acker (via the -users mailing list) Patch by: Corydon76 (license 14) 2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson * apps/app_queue.c: Prevent a potential three-thread deadlock. Also added a comment block to explicitly state the locking order necessary inside app_queue. (closes issue #11862) Reported by: flujan Patches: 11862.patch uploaded by putnopvut (license 60) Tested by: flujan * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl fails for Zaptel timing device. (closes issue #11891) Reported by: tzafrir 2008-02-08 15:08 +0000 [r102968] Joshua Colp * channels/chan_iax2.c: Make sure the presence of dbsecret is factored into user scoring. (closes issue #11952) Reported by: bbhoss 2008-02-07 19:53 +0000 [r102858] Jason Parker * res/res_features.c: Specify which digit string was matched in debug message. (closes issue #11949) Reported by: dimas Patches: v1-feature-debug.patch uploaded by dimas (license 88) 2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming * configs/zapata.conf.sample: document usage of 'transfer' configuration option for ISDN PRI switch-side transfers 2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp * channels/chan_sip.c: Only consider a T.38-only INVITE compatible if we have both a joint capability between us and them and if they provided T.38. * main/global_datastores.c: Add missing header file and ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by: snuffy 2008-02-06 15:19 +0000 [r102651] Russell Bryant * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so that it gets inherited by outbound channels. (due to a discussion between me and a user via email) 2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming * pbx/Makefile, res/Makefile: ensure that all remaining multi-object modules are built using their proper CFLAGS and include directory paths 2008-02-06 00:26 +0000 [r102576] Tilghman Lesher * apps/app_voicemail.c: Move around some defines to unbreak ODBC storage. (closes issue #11932) Reported by: snuffy 2008-02-05 20:02 +0000 [r102453] Mark Michelson * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes issue #11919) Reported by: eferro Patches: mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) Tested by: eferro 2008-02-05 19:52 +0000 [r102450] Joshua Colp * channels/chan_sip.c: If a REGISTER attempt comes in that is a retransmission of a previous REGISTER do not create a new nonce value. (issue #BE-381) 2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming * channels/Makefile: ensure that components of chan_misdn.so are built using any special build options that the configure script generated (reported by Philipp Kempgen on asterisk-dev) 2008-02-05 15:09 +0000 [r102378] Joshua Colp * res/res_clioriginate.c: Perform dialing asynchronously when using the originate CLI command so the CLI does not appear to block. (closes issue #11927) Reported by: bbhoss 2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher * main/asterisk.c, utils/muted.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Cross-platform fix: OS X now deprecates the use of the daemon(3) API. (closes issue #11908) Reported by: oej Patches: 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 * funcs/func_strings.c: Missing braces. (closes issue #11912) Reported by: dimas Patches: sprintf.patch uploaded by dimas (license 88) 2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE (according to RFC 3261) (closes issue #9492) Reported by: kryptolus Patches: bug9492.txt uploaded by oej (license 306) Tested by: oej * channels/chan_sip.c: Handle ACK and CANCEL in an invite transaction - even if we get INFO transactions during the actual call setup. (closes issue #10567) Reported by: jacksch Tested by: oej Patch by: oej inspired by suggestions from neutrino88 in the bug tracker 2008-02-01 23:06 +0000 [r101989] Russell Bryant * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz, it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but people follow it anyway, because it's the spec ...) 2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded voicemail, and fixed several other bugs while I'm in the area. (closes issue #11615) Reported by: jamessan Patches: 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, jamessan * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Change detection of getifaddrs to use AST_C_COMPILE_CHECK, backported from trunk (as suggested by kpfleming) 2008-02-01 17:41 +0000 [r101822] Jason Parker * apps/app_authenticate.c: Remove a needless (and incorrect) call to feof() after fgets(). This would have exited the loop early if you had an authentication file with no newline at the end. 2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant * apps/app_authenticate.c: off by one error * apps/app_authenticate.c: Don't overwrite the last character of a line if it's not a newline. This would happen if the last line in the file doesn't have a newline. (pointed out by Qwell) 2008-02-01 15:55 +0000 [r101772] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, configure.ac, main/acl.c: Compatibility fix for OpenWRT (reported by Brian Capouch via the mailing list) 2008-02-01 00:32 +0000 [r101693] Russell Bryant * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial strings for the case that no peer or hostname was provided, which is the one part of the dial string that is absolutely required. If it's not there, bail out. (closes issue #11897) Reported by sokhapkin Patch by me 2008-02-01 00:06 +0000 [r101649] Mark Michelson * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle periods of no channel activity" (closes issue #9256) Reported by: cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, rjain 2008-01-31 Russell Bryant * Asterisk 1.4.18 released. 2008-01-31 23:10 +0000 [r101601] Russell Bryant * main/translate.c, main/file.c: Fix a couple of places where ast_frfree() was not called on a frame that came from a translator. This showed itself by g729 decoders not getting released. Since the flag inside the translator frame never got unset by freeing the frame to indicate it was no longer in use, the translators never got destroyed, and thus the g729 licenses were not released. (closes issue #11892) Reported by: xrg Patches: 11892.diff uploaded by russell (license 2) Tested by: xrg, russell 2008-01-31 21:00 +0000 [r101531] Mark Michelson * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the beginning of an absolute pathname. 2. If ast_monitor_change_fname is called and the new filename is the same as the old, then exit early and don't set the filename_changed field in the monitor structure. Setting it in this case was causing ast_monitor_stop to erroneously delete them. (closes issue #11741) Reported by: garlew Tested by: putnopvut 2008-01-31 19:52 +0000 [r101482] Jason Parker * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes for struct in_addr funkiness. Issue #11885, patch by snuffy. 2008-01-31 19:30 +0000 [r101480] Steve Murphy * main/pbx.c: closes issue #11845; that's the one where there's a 1004 byte cdr leak with every AMI Redirect to a zap channel 2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant * channels/chan_agent.c: Add more missing locking of the agents list ... * channels/chan_agent.c: Move the locking from find_agent() into the agent dialplan function handler to ensure that the agent doesn't disappear while we're looking at it. * channels/chan_agent.c: Add missing locking to the find_agent() function. 2008-01-30 15:41 +0000 [r101222] Joshua Colp * main/slinfactory.c: Fix an issue where if a frame of higher sample size preceeded a frame of lower sample size and ast_slinfactory_read was called with a sample size of the combined values or higher a crash would happen. (closes issue #11878) Reported by: stuarth 2008-01-30 15:34 +0000 [r101219] Jason Parker * configs/extensions.conf.sample: Change default config to use descending channel order of groups, rather than ascending. Fixes a potential source of confusion in glare-type situations. Issue 11875, reported by JimVanM. 2008-01-30 15:23 +0000 [r101216] Mark Michelson * apps/app_queue.c: Fix a logic error with regards to autofill. Prior to this change, it was possible for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting to call a member. This change fixes this. 2008-01-30 11:20 +0000 [r101152] Olle Johansson * channels/chan_sip.c: Stop musiconhold on attended transfer. (closes issue #11872) Reported by: gareth Patches: svn-101018.patch uploaded by gareth (license 208) 2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard * build_tools/make_version: updated build_tools to handle the autotag directory structure changes; changes related to BE-353. Patch by The Russell and reviewed by The Me. 2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson * apps/app_queue.c: Remove a memory leak from updating realtime queues * apps/app_queue.c: Fixing an erroneous return value returned when attempting to pause or unpause a queue member fails. Fixes BE-366, thanks to John Bigelow for writing the patch. 2008-01-29 17:57 +0000 [r100934] Joshua Colp * apps/app_mixmonitor.c: Don't forget to record the channel so we know whether it is bridged or not later. (closes issue #11811) Reported by: slavon 2008-01-29 17:43 +0000 [r100932] Russell Bryant * main/Makefile: Fix the last couple of issues related to building from a path that contains spaces. (closes issue #11834) 2008-01-29 17:41 +0000 [r100930] Jason Parker * channels/misdn_config.c: Initialize an array to 0s if config option not specified. (closes issue #11860) Patches: misdn_get_config.v1.diff uploaded by IgorG (license 20) 2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant * Makefile: Use GNU make magic instead of shell magic to escape spaces in the working directory. (related to issue #11834) * Makefile: Fix building Asterisk when the working path has spaces in it. (closes issue #11834) Reported by: spendergrass Patched by: me 2008-01-29 16:10 +0000 [r100835] Jason Parker * channels/chan_zap.c: Allow zap groups above 30 to work properly. (closes issue #11590) Reported by: tbsky 2008-01-29 10:36 +0000 [r100793] Christian Richter * channels/chan_misdn.c: fixed potential segfault in misdn show channels CLI command 2008-01-29 08:26 +0000 [r100740] Olle Johansson * channels/chan_sip.c: (closes issue #11736) Reported by: MVF Patches: bug11736-2.diff uploaded by oej (license 306) Tested by: oej, MVF, revolution (russellb: This was the showstopper for the release.) 2008-01-28 21:02 +0000 [r100675] Tilghman Lesher * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly (went to 't' instead of 'T') 2008-01-28 20:55 +0000 [r100673] Mark Michelson * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of chan_vpb. It is alive and well. 2008-01-28 20:42 +0000 [r100672] Jason Parker * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put greeting files into the database like we do with the others. Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded by dimas (license 88) 2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant * channels/chan_sip.c: For some reason, the use of this strdupa() is leading to memory corruption on freebsd sparc64. This trivial workaround fixes it. (closes issue #10300, closes issue #11857, reported by mattias04 and Home-of-the-Brave) * res/res_features.c: Fix a crash in ast_masq_park_call() (issue #11342) Reported by: DEA Patches: res_features-park.txt uploaded by DEA (license 3) 2008-01-28 18:23 +0000 [r100624] Jason Parker * channels/chan_zap.c: Correct a comment which made little/no sense. 2008-01-28 17:15 +0000 [r100581] Russell Bryant * main/channel.c, channels/chan_local.c, include/asterisk/channel.h: Make some deadlock related fixes. These bugs were discovered and reported internally at Digium by Steve Pitts. - Fix up chan_local to ensure that the channel lock is held before the local pvt lock. - Don't hold the channel lock when executing the timing function, as it can cause a deadlock when using chan_local. This actually changes the code back to be how it was before the change for issue #10765. But, I added some other locking that I think will prevent the problem reported there, as well. 2008-01-27 21:59 +0000 [r100465] Tilghman Lesher * main/rtp.c, channels/chan_mgcp.c, main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c, channels/chan_h323.c, include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from the scheduler, ignoring the return value could possibly cause memory to be accessed after it is freed, which causes all sorts of random memory corruption. Instead, if a deletion fails, wait a bit and try again (noting that another thread could change our taskid value). (closes issue #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan, stuarth` 2008-01-25 22:32 +0000 [r100418] Mark Michelson * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is now preferred that users of Voicetronix products use chan_zap in combination with their zaptel drivers. 2008-01-25 21:24 +0000 [r100378] Jason Parker * channels/chan_sip.c: This would have never been true, since we're passing (sizeof(req.data) - 1) as the len to recvfrom(). 2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming * include/asterisk/app.h: make these macros not assume that the only other field in the structure is 'argc'... this is true when someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable to define your own structure as long as it has the right fields 2008-01-24 17:22 +0000 [r100164] Russell Bryant * main/asterisk.c: Update main Asterisk copyright info to 2008 2008-01-24 16:41 +0000 [r100138] Jason Parker * main/acl.c: Fix compilation on Solaris. (closes issue #11832) Patches: bug-11832.diff uploaded by snuffy (license 35) 2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson * channels/chan_sip.c: Second attempt. Don't change invitestate when receiving 18x messages in CANCEL state. (issue #11736) Reported by: MVF Patch by oej. * channels/chan_sip.c: Make sure we don't cancel destruction on calls in CANCEL state, even if we get 183 while waiting for answer on our CANCEL. (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by oej (license 306) Tested by: MVF 2008-01-23 20:25 +0000 [r99975] Mark Michelson * apps/app_externalivr.c: Fixing a typo. 2008-01-23 17:46 +0000 [r99923] Russell Bryant * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the beginning of a list of channels to potentially spy on. However, if there were no matching channels, it would beep at you over and over, which is pretty annoying. Now, it will only beep once in the case that there are no channels to spy on, but it will still beep again once it reaches the beginning of the channel list again. (closes issue #11738, patched by me) 2008-01-23 16:18 +0000 [r99878] Mark Michelson * channels/chan_sip.c: These flag tests were illogical. They were testing sip_peer flags on a sip_pvt. Thanks to Russell for helping to get this odd problem figured out. 2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher * apps/app_voicemail.c: When we reset the password via an external command, we should also reset the password stored in the in-memory list, too (otherwise it doesn't really take effect). (closes issue #11809) Reported by: davetroy Patches: fix_externpass.diff uploaded by davetroy (license 384) * res/res_odbc.c: Oops, should have checked for a NULL obj, here, too * main/acl.c: Just confirmed that all current platforms need this header file 2008-01-22 20:56 +0000 [r99652] Olle Johansson * channels/chan_sip.c: Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old head to avoid to heavy memory allocations on some systems. 2008-01-22 20:34 +0000 [r99643] Tilghman Lesher * main/acl.c: Fix the defines for OS X (and Solaris, too) 2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson * channels/chan_local.c, res/res_features.c, channels/chan_agent.c, apps/app_followme.c: Add more dependencies on chan_local and add a note to the description of chan_local so that people don't disable it in menuselect just to clean up. * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial still runs without chan_local, but will be missing forwarding functionality. 2008-01-22 16:54 +0000 [r99540] Tilghman Lesher * main/acl.c: Ensure that we can get an address even when we don't have a default route. (closes issue #9225) Reported by: junky Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 (license 14) Tested by: oej, loloski, sergee 2008-01-22 15:08 +0000 [r99501] Olle Johansson * channels/chan_sip.c: Cleaning up some documentation that led to confusion in a bug report 2008-01-21 23:55 +0000 [r99426] Mark Michelson * channels/chan_local.c: Fixing an issue wherein monitoring local channels was not possible. During a channel masquerade, the monitors on the two channels involved are swapped. In 99% of the cases this results in the desired effect. However, if monitoring a local channel, this caused the monitor which was on the local channel to get moved onto a channel which is immediately hung up after the masquerade has completed. By swapping the monitors prior to the masquerade, we avoid the problem by tricking the masquerade into placing the monitor back onto the channel where we want it. During the investigation of the issue, the channel's monitor was the only thing that was swapped in such a manner which did not make sense to have done. All other variable swapping made sense. 2008-01-21 18:11 +0000 [r99341] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample, include/asterisk/res_odbc.h: Permit the user to specify number of seconds that a connection may remain idle, which fixes a crash on reconnect with the MyODBC driver. (closes issue #11798) Reported by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak 2008-01-21 16:01 +0000 [r99301] Joshua Colp * channels/chan_sip.c: Bump the buffer size for Via headers up to 512. There are some exceptionally large Via headers out there. (closes issue #11783) Reported by: ofirroval 2008-01-19 10:05 +0000 [r99187] Russell Bryant * main/slinfactory.c: Fix a couple of memory leaks with frame handling. Specifically, ast_frame_free() needed to be called on the frame that came from the translator to signed linear. 2008-01-18 22:57 +0000 [r99127] Joshua Colp * include/asterisk/channel.h: Remove the __ in front of the unused variable. This causes some compilers to freak out. 2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant * include/asterisk/translate.h, main/frame.c: Revert adding the packed attribute, as it really doesn't make sense why that would do any good. Fix the real bug, which is to do the check to see if the frame came from a translator at the beginning of ast_frame_free(), instead of at the end. This ensures that it always gets checked, even if none of the parts of the frame are malloc'd, and also ensures that we aren't looking at free'd memory in the case that it is a malloc'd frame. (closes issue #11792, reported by explidous, patched by me) * include/asterisk/translate.h: Since we're relying on the offset between the frame and the beginning of the translator pvt struct, set the packed attribute to make sure we get to the right place. (potential fix for issue #11792) 2008-01-18 17:13 +0000 [r99032] Terry Wilson * res/res_features.c: This should at least temporarily fix a problem where the 't' Dial option is incorrectly passed to the transferee when built-in attended transfers are used. There is still a problem with 'T', but better to fix some problems than no problems while we work on it. (closes issue #7904) Reported by: k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee (license 138) Tested by: sergee, otherwiseguy 2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni * configs/cdr.conf.sample: doh! revert a revert of a revert (changed by mistake in 99010) * main/manager.c, configs/cdr.conf.sample: missed that one while reverting * main/manager.c: reverting 99001 - We need the Max-Age for extending the life of cookie mansession_id 2008-01-17 22:37 +0000 [r99004] Russell Bryant * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: Have IAX2 optimize the codec translation path just like chan_sip does it. If the caller's codec is in our codec list, move it to the top to avoid transcoding. (closes issue #10500) Reported by: stevedavies Patches: iax-prefer-current-codec.patch uploaded by stevedavies (license 184) iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184) Tested by: stevedavies, pj, sheldonh 2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming * main/manager.c: we should only send the Set-Cookie header to the browser on the first response after creating a manager session, not on every response (doing so causes the browser to clear any local cookies it may have associated with the session) 2008-01-17 16:19 +0000 [r98991] Jason Parker * configs/zapata.conf.sample: Add a clarification about the immediate= option of zapata.conf Issue 11784, patch by klaus3000. 2008-01-16 22:36 +0000 [r98982] Russell Bryant * .cleancount, include/asterisk/channel.h: Add an unused pointer to the ast_channel struct. This makes the ast_channel structure retain the same size as it had in previous 1.4 releases. Also, all of the offsets for members in the structure are still the same (except for the two pointers that got replaced for the new spy/whisper architecture.) 2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp * .cleancount: Bump up cleancount due to previous commit that changed the channel structure. * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c, main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h (added), main/Makefile, include/asterisk/chanspy.h (removed), include/asterisk/channel.h, main/audiohook.c (added): Replace current spy architecture with backport of audiohooks. This should take care of current known spy issues. * channels/chan_iax2.c: Add missing NULLs at end of two ast_load_realtimes. (closes issue #11769) Reported by: tequ Patches: chaniax.patch uploaded by dimas (license 88) 2008-01-16 17:20 +0000 [r98964] Mark Michelson * channels/chan_local.c: Fix a deadlock in chan_local in local_hangup. There was contention because the local_pvt was held and it was attempting to lock a channel, which is the incorrect locking order. (closes issue #11730) Reported by: UDI-Doug Patches: 11730.patch uploaded by putnopvut (license 60) Tested by: UDI-Doug 2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp * main/dial.c: Introduce a lock into the dialing API that protects it when destroying the structure. (closes issue #11687) Reported by: callguy Patches: 11687.diff uploaded by file (license 11) * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes issue #11777) Reported by: tootai * channels/chan_sip.c: Don't drop the old record route information when dealing with packets related to a reinvite. (closes issue #11545) Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by kebl0155 (license 356) * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, codecs/codec_speex.c, configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex. (closes issue #11693) Reported by: yzg 2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant * channels/chan_sip.c: Change a buffer in check_auth() to be a thread local dynamically allocated buffer, instead of a massive buffer on the stack. This fixes a crash reported by Qwell due to running out of stack space when building with LOW_MEMORY defined. On a very related note, the usage of BUFSIZ in various places in chan_sip is arbitrary and careless. BUFSIZ is a system specific define. On my machine, it is 8192, but by definition (according to google) could be as small as 256. So, this buffer in check_auth was 16 kB. We don't even support SIP messages larger than 4 kB! Further usage of this define should be avoided, unless it is used in the proper context. * main/rtp.c, include/asterisk/translate.h, main/frame.c, main/translate.c, main/abstract_jb.c, channels/chan_iax2.c, codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... 2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp * channels/chan_sip.c: Based on the boundary found move over the correct amount. (closes issue #11750) Reported by: tasker * channels/chan_sip.c: Accept "; boundary=" not just ";boundary=" in the multipart mixed content type. (closes issue #11750) Reported by: tasker 2008-01-14 20:59 +0000 [r98849] Mark Michelson * apps/app_voicemail.c: Adding in appropriate unlocks for the locks I added. Thanks to joetester on IRC for pointing this out. 2008-01-14 17:38 +0000 [r98774] Russell Bryant * main/translate.c: Revert a change that introduces an unacceptable performance hit and is causing memory leaks ... (from rev 97973) 2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson * apps/app_queue.c: Fixing another compilation error. I'm a bit off today :( * apps/app_queue.c: Oops. Last commit had compilation error. * apps/app_queue.c: Adding explicit defaults for missing options to init_queue. This is necessary because if a user either removes or comments one of these options and reloads their queues, the option will not reset to its default, instead maintaining the value from prior to the reload. Thanks to John Bigelow for pointing this error out to me. 2008-01-12 00:05 +0000 [r98467] Tilghman Lesher * res/res_odbc.c: Add a connection timeout attribute, as that was what was intended with the login timeout, but ODBC divides it up into 2 different timeouts. (Closes issue #11745) 2008-01-11 22:46 +0000 [r98390] Russell Bryant * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems. (closes issue #11646) Reported by: caio1982 Patches: dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, mvanbaak 2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni * main/http.c: Comment explaining how to force browser to always read some html files from server. 2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp * main/rtp.c: If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) * res/res_agi.c: If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else. (closes issue #11743) Reported by: davevg Patches: res_agi.diff uploaded by davevg (license 209) 2008-01-11 19:10 +0000 [r98315] Mark Michelson * main/channel.c: Properly report the hangup cause as no answer when someone does not answer (closes issue #10574, reported by boch, patched by moy) 2008-01-11 18:25 +0000 [r98266] Tilghman Lesher * codecs/gsm/Makefile: Add another exception (which doesn't work) for -march optimization flag. Reported by: thomasmebes Patch by: tilghman (Closes issue #11563) 2008-01-11 18:25 +0000 [r98265] Russell Bryant * doc/security.txt, main/asterisk.c, configure, include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, makeopts.in: Backport the ability to set the ToS bits on Linux when not running as root. Normally, we would not backport features into 1.4, but, I was convinced by the justification supplied by the supplier of this patch. He pointed out that this patch removes a requirement for running as root, thus reducing the potential impacts of security issues. (closes issue #11742) Reported by: paravoid Patches: libcap.diff uploaded by paravoid (license 200) 2008-01-11 17:22 +0000 [r98219] Joshua Colp * apps/app_followme.c: Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue #10327) Reported by: kkiely 2008-01-11 15:52 +0000 [r98164] Tilghman Lesher * channels/chan_sip.c: Back out changes from revision 97077, since it wasn't perfect 2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant * main/frame.c: Fix samples vs. length calculations for g722 * main/translate.c: Simplify this code with a suggestion from Luigi on the asterisk-dev list. Instead of using is16kHz(), implement a format_rate() function. * main/translate.c: Fix various timing calculations that made assumptions that the audio being processed was at a sample rate of 8 kHz. 2008-01-10 23:08 +0000 [r97973] Tilghman Lesher * channels/chan_sip.c, main/translate.c: 1) When we get a translated frame out, clone it, because if the translator pvt is freed before we use the frame, bad things happen. 2) Getting a failure from ast_sched_delete means that the schedule ID is currently running. Don't just ignore it. (Closes issue #11698) 2008-01-10 21:57 +0000 [r97925] Mark Michelson * apps/app_voicemail.c: Let us leave a voicemail for ourself if we have logged into VoiceMailMain and chosen to leave a message. (closes issue #11735, reported and patched by jamessan) 2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro. * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least. 2008-01-10 20:12 +0000 [r97847] Jason Parker * include/asterisk/frame.h: Fix a comment that is no longer true. 2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample, pbx/kdeconsole_main.cc (removed): Remove other remnants of pbx_kdeconsole * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't worked in ages, and nobody has complained. (closes issue #11706, reported by caio1982) 2008-01-10 15:07 +0000 [r97697] Joshua Colp * funcs/func_groupcount.c: Don't try to copy the category from the group if no category exists. (closes issue #11724) Reported by: IgorG Patches: group_count.v1.patch uploaded by IgorG (license 20) 2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose messages * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't actually load on my system still (related to issue #11706) 2008-01-09 20:28 +0000 [r97618-97622] Jason Parker * main/cli.c: Correctly display a message if a command could not be found. Also fix a comment which may have led to this happening. Issue 11718, reported by kshumard. * main/cli.c: Fix some locking and return value funkiness. We really shouldn't be unlocking this lock inside of a function, unless we locked it there too. 2008-01-09 18:48 +0000 [r97575] Mark Michelson * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some smaller functions this time 2008-01-09 18:02 +0000 [r97529] Russell Bryant * res/res_features.c: Fix saying the parking space number to the caller doing the parking ... 2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming * codecs/codec_zap.c: report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found 2008-01-09 16:44 +0000 [r97489] Philippe Sultan * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. 2008-01-09 16:11 +0000 [r97450] Joshua Colp * apps/app_meetme.c: Don't do conferencing totally in Zaptel if Monitor is running on the channel. (closes issue #11709) Reported by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license 371) 2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming * channels/chan_zap.c: pass the right variable to get an error string... oops * channels/chan_zap.c: add error number output to ioctl failure messages to help with debugging 2008-01-09 00:44 +0000 [r97350] Tilghman Lesher * main/cli.c, main/editline/readline.c: Allow filename completion on zero-length modules, remove a memory leak, remove a file descriptor leak, and make filename completion thread-safe. Patched and tested by tilghman. (Closes issue #11681) 2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson * apps/app_queue.c: use the \retval doxygen command properly * apps/app_queue.c: Part 1 of N of adding doxygen comments to app_queue. I picked some of the most common functions used (which also happen to be some the biggest/ugliest functions too) to document first. I'm pretty new to doxygen so criticism is welcome. * apps/app_queue.c: Some coding guidelines-related cleanup 2008-01-08 20:48 +0000 [r97195] Joshua Colp * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp. (closes issue #11443) Reported by: eferro Patches: dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license 337) 2008-01-08 20:47 +0000 [r97194] Tilghman Lesher * main/autoservice.c, main/utils.c: Increase constants to where we're less likely to hit them while debugging. (Closes issue #11694) 2008-01-08 20:42 +0000 [r97192] Mark Michelson * apps/app_voicemail.c: Making some changes designed to not allow for a corrupted mailstream for a vm_state. 1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs. 2. Make sure to always grab the persistent vm_state when mailstream access is necessary. 3. Correct an incorrect return value in the init_mailstream function. (closes issue #11304, reported by dwhite) 2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp * funcs/func_groupcount.c: If no group has been provided to the GROUP_COUNT dialplan function then use the first one specific to the channel. (closes issue #11077) Reported by: m4him * apps/app_queue.c: Make app_queue calls work with directed pickup. (closes issue #11700) Reported by: jbauer 2008-01-08 18:02 +0000 [r97077] Tilghman Lesher * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes, found in issue #11386, but not completely closing that issue. 2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant * configs/extensions.conf.sample, /: Merged revisions 96931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com ........ * res/res_smdi.c: Don't crash if something happens when setting up an SMDI interface and it gets destroyed before the SMDI port handling thread gets created. 2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan * res/res_jabber.c: Indentation fix, makes the code easier to read * res/res_jabber.c: Compute the base64 value over the [authzid]\0authcid\0password string, thus excluding the trailing NULL byte. This change has already been committed to trunk, see #11644. 2008-01-05 02:09 +0000 [r96644] Russell Bryant * main/devicestate.c: Don't pass an empty string as the device name. 2008-01-04 23:03 +0000 [r96575] Tilghman Lesher * main/devicestate.c: Fix the problem of notification of a device state change to a device with a '-' in the name. Could probably do with a better fix in trunk, but this bug has been open way too long without a better solution. Reported by: stevedavies Patch by: tilghman (Closes issue #9668) 2008-01-04 22:55 +0000 [r96573] Jason Parker * res/res_features.c: Properly continue in the dialplan if using PARKINGEXTEN and the slot is full. Issue 11237, patch by me. 2008-01-04 19:27 +0000 [r96525] Tilghman Lesher * channels/chan_sip.c: If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie. Reported and patched by: one47 (Closes issue #11535) 2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant * channels/chan_zap.c: Make use of the temporary channel pointer while the pvt is unlocked. (closes issue #11675) Reported by: flefoll Patches: chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244) * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has been destroyed before we get to this point (closes issue #11672, reported by snuffy, patched by me) 2008-01-03 21:37 +0000 [r96318] Tilghman Lesher * res/res_config_pgsql.c: Missed initialization caused crash. Reported and fixed by: tiziano (Closes issue #11671) 2008-01-03 12:12 +0000 [r96198-96199] Christian Richter * channels/chan_misdn.c: make sure frame is completely clean, before we send it to asterisk as DTMF. If we don't make it clean, it happens that one way audio occurs.. * channels/chan_misdn.c: when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense. 2008-01-02 23:46 +0000 [r96102] Mark Michelson * apps/app_queue.c: We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that multiple members can have the same name, since the variable was not reset on each iteration of the loop. 2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant * pbx/pbx_config.c: Convert locks of the contexts list in pbx_config to the appropriate rdlock or wrlock * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts list. * apps/app_macro.c: app_macro only needs a rdlock on the contexts list. 2008-01-02 Russell Bryant * Asterisk 1.4.17 released. 2008-01-02 20:24 +0000 [r95946] Joshua Colp * channels/chan_sip.c: Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-028) (closes issue #11637) Reported by: greyvoip 2008-01-02 17:51 +0000 [r95890] Mark Michelson * apps/app_queue.c: A change to improve the accuracy of queue logging in the case where a member does not answer during the specified timeout period. Prior to this change, there was a small chance that the member name recorded in this case would be blank. Also prior to this change, if using the ringall strategy, if no one answered the call during the specified timeout, the member name listed in the queue log would randomly be one of the members that was rung. (closes issue #11498, reported and tested by hloubser, patched by me) 2007-12-31 23:43 +0000 [r95577] Mark Michelson * main/pbx.c: Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension into ast_merge_contexts_and_delete (sans the extra lock). (this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the problematic area experienced by the reporters of that issue) 2007-12-31 20:27 +0000 [r95470] Tilghman Lesher * funcs/func_env.c: Allow the default "0" to be returned if the STAT fails (Closes issue #11659) 2007-12-28 18:24 +0000 [r95191] Russell Bryant * channels/chan_sip.c: Remove duplicate increment of the header count in the add_header() function. (closes issue #11648) Reported by: makoto Patch provided by sergee, committed patch by me, inspired by comments from putnopvut 2007-12-28 00:16 +0000 [r95095] Mark Michelson * apps/app_queue.c: I found a bug while browsing the queue code and managed to reproduce it in a small setup. If a queue uses the ringall strategy, it was possible through unfortunate coincidence for a single member at a given penalty level to make app_queue think that all members at that penalty level were unavailable and cause the members at the next penalty level to be rung. With this patch, we will only move to the next penalty level if ALL the members at a given penalty level are unreachable. 2007-12-27 21:40 +0000 [r95024] Russell Bryant * main/channel.c: Don't report a syntax error when an empty string is passed to ast_get_group. Just return 0. (closes issue #11540) Reported by: tzafrir Patches: group_empty.diff uploaded by tzafrir (license 46) -- slightly changed by me 2007-12-27 20:09 +0000 [r94977] Mark Michelson * main/io.c: Fixing a typo in a comment. 2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp * channels/chan_h323.c: Include types.h in chan_h323 as without it it can not be compiled on some operating systems like FreeBSD to name one. (closes issue #11585) Reported by: sobomax Patches: chan_h323.c.diff uploaded by sobomax (license 359) * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is set or not on the dialog. It is possible for it to be a pointer to NULL. (closes issue #11557) Reported by: FuriousGeorge 2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant * main/pbx.c: Now that the contexts lock is a read/write lock, it should not be locked here in ast_hint_state_changed(). This makes it get locked recursively which now causes a deadlock. (closes issue #11080, thanks to callguy for the access to a deadlocked machine) * include/asterisk/translate.h, main/translate.c: Use the constant that I really meant to use here ... * main/translate.c: Change ast_translator_best_choice() to only pay attention to audio formats. This fixes a problem where Asterisk claims that a translation path can not be found for channels involving video. (closes issue #11638) Reported by: cwhuang Tested by: cwhuang Patch suggested by cwhuang, with some additional changes by me. 2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming * main/manager.c: make this comment explain the situation in an even more explicit fashion 2007-12-26 20:43 +0000 [r94808] Tilghman Lesher * main/manager.c: Workaround for what is probably a glibc bug (but we'll see this crop up again and again, if we don't add the workaround). Reported by: rolek Patch by: tilghman (Closes issue #11601, closes issue #11426) 2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting autoservice, remember it and ensure that the channel has the same setting when autoservice gets stopped. (pointed out by d1mas, patched up by me) * main/autoservice.c: When a channel is in autoservice, mark a flag on the channel that says that we only care about the END of a digit. That way, no magic digit emulation stuff will happen when all we're doing is queueing up END frames. * res/res_features.c: Don't try to send a parked call back to itself. (closes issue #11622, reported by djrodman, patched by me) * main/autoservice.c: Don't store DTMF BEGIN frames while a channel is in autoservice. It's just going to make ast_read() do a lot of extra work when the channel comes back out of autoservice. (closes issue #11628, patched by me) * Makefile: List include/asterisk/version.h as a .PHONY target because we want the commands listed for this target to be executed regardless of whether the file exists or not. This fixes having the version not up to date when running from svn. (closes issue #11619, reported by plack, fixed by me) 2007-12-25 02:27 +0000 [r94769] Joshua Colp * channels/chan_sip.c: file says... build on the builders. 2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher * main/channel.c: Race: we need to wait to queue a NewChannel event until after the channel is inserted into the channel list. The reason is because some manager users immediately queue requests from the channel when they see that event and are confused when Asterisk reports no such channel. (Closes issue #11632) * channels/chan_sip.c: More deadlock avoidance code (this time between sip_monitor and sip_hangup) Reported by: apsaras Patch by: tilghman (Closes issue #11413) * channels/chan_sip.c: Another bit of bad logic in realtime_peer Reported by: dimas Patch by: dimas (Closes issue #11631) 2007-12-23 01:21 +0000 [r94660] Tilghman Lesher * channels/chan_sip.c: Argh... I suppose third time's the charm. 2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson * apps/app_voicemail.c: Bunch of coding guidelines cleanup * apps/app_voicemail.c: Better quota support for using IMAP storage voicemail (closes issue #11415, reported by jaroth) (closes issue #11152, reported by selsky) Patch provided by jaroth * apps/app_voicemail.c: The mail_copy c-client function does not expect a full imap mailbox string, just the name of the mailbox. (closes issue #11419, reported and patched by jaroth, with additional patchwork from me) * main/dial.c: Since we are freeing list elements within a list traversal, we need to use the safe traversal and remove the item from the list before freeing it. (closes issue 11612, reported by dtyoo) 2007-12-21 16:37 +0000 [r94466] Russell Bryant * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to a read/write lock to resolve a deadlock. This has a nice side benefit of improving performance. :) (closes issue #11609) (closes issue #11080) 2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson * apps/app_queue.c: Removing a debug message I accidentally just committed * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying dates and times. Also some coding guidelines cleanup. (closes issue #11599, reported and patched by caio1982, coding guidelines cleanup by me) 2007-12-21 15:07 +0000 [r94418] Tilghman Lesher * main/asterisk.c: Fix for restart-as-user problem reported via the -dev list 2007-12-20 Russell Bryant * Asterisk 1.4.16.2 released. 2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) | 5 lines Fix another potential seg fault ... (closes issue #11606) Reported by: dimas ........ * channels/chan_zap.c: Fix a deadlock in d-channel handling in chan_zap. This deadlock was introduced by the fix to ensure that channels are properly locked when handling channel variables. There were sections of this code where the channel pvt was locked before the channel lock, when in fact it _must_ be the other way around. (closes issue #11582) Reported by: bugi * /: Blocked revisions 94214 via svnmerge ........ r94214 | russell | 2007-12-20 11:29:11 -0600 (Thu, 20 Dec 2007) | 2 lines Fix a couple of places where it's possible to dereference a NULL pointer. ........ 2007-12-19 23:02 +0000 [r94122] Mark Michelson * res/res_monitor.c: Sox versions 13.0.0 and newer do not have "soxmix" and instead use sox -m. res_monitor needs to use this if the user does not have soxmix. (closes issue #11589, reported by amessina, patch inspired by amessina but with a flourish from me) 2007-12-19 22:48 +0000 [r94077] Russell Bryant * configure, include/asterisk/autoconfig.h.in, configure.ac: Check for the existence of the soxmix application on the target platform and have the result available in autoconfig.h. (part of issue #11589) 2007-12-19 Russell Bryant * Asterisk 1.4.16.1 released. 2007-12-19 17:29 +0000 [r93955] Joshua Colp * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is NULL. 2007-12-19 17:04 +0000 [r93949] Tilghman Lesher * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't defined (Closes issue #11602) 2007-12-18 22:42 +0000 [r93764] Jason Parker * channels/chan_skinny.c: FreeBSD also does not have byte swap functions. Issue 11586, patch by sobomax. 2007-12-18 Russell Bryant * Asterisk 1.4.16 released. 2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 93667 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........ 2007-12-18 17:02 +0000 [r93625] Mark Michelson * main/channel.c: Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored had one audio file recorded and one empty audio file saved. (closes issue #11529, reported by atis patched by me) 2007-12-17 22:56 +0000 [r93381-93420] Jason Parker * main/translate.c: What was I thinking when I wrote this masterpiece? -1 + 1 = 0.. who woulda thunk it?. 2007-12-17 22:28 +0000 [r93377] Joshua Colp * main/utils.c: Do not try to access information about a lock when printing out a trylock attempt. It is possible for the lock that it references to no longer be valid. This would have caused segfaults or deadlocks. (issue #BE-263) (closes issue #11080) Reported by: callguy (closes issue #11100) Reported by: callguy 2007-12-17 21:12 +0000 [r93336] Tilghman Lesher * include/asterisk/time.h: Today is tomorrow's yesterday, and yesterday's tomorrow is today, and tomorrow's tomorrow is the day after tomorrow, so who cares if you recycle anyway? If this confuses you, that's nothing compared to what this fixes. ;-) 2007-12-17 19:53 +0000 [r93291] Mark Michelson * apps/app_voicemail.c: We need to create the directory for a voicemail user even if they are using IMAP storage since greetings are stored in the filesystem. (closes issue #11388, reported by spditner, patch by me inspired by a patch by spditner) 2007-12-17 18:05 +0000 [r93250] Joshua Colp * channels/chan_zap.c: If a call is received with a called number IE containing nothing go to the 's' extension. (closes issue #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by Corydon76 (license 14) 2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile: fix some copy-and-paste leftovers 2007-12-17 07:15 +0000 [r93182] Olle Johansson * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add dependencies on res_monitor and res_features. I wonder if Asterisk can run at all without res_features. My guess is that there's propably a lot of more modules and the core that depends on it. Reported by: caio1982 (closes issue #11574) 2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming * formats, Makefile, codecs/Makefile, funcs, apps/Makefile, configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in, formats/Makefile, pbx, res, channels, funcs/Makefile, codecs, include/asterisk/autoconfig.h.in, build_tools/make_version, apps, configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. 2007-12-14 17:36 +0000 [r93000] Russell Bryant * main/config.c: There are a lot of existing systems that #include non-existent files. So, to make the transition to treating this as an error a bit less painless, just issue a huge error message for now. Then, later, we can reinstate the code that treats it as a failure. (Thanks to philippel for the feedback) 2007-12-14 15:16 +0000 [r92937] Joshua Colp * channels/chan_sip.c: Up the length of the format on the SIP channel since it can now be rather long. (closes issue #11552) Reported by: francesco_r 2007-12-14 15:05 +0000 [r92934] Christian Richter * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS state setting and overlap_task thread starting. 2007-12-14 15:01 +0000 [r92933] Tilghman Lesher * res/res_agi.c: Change help documentation to match actual behavior (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman (Closes issue #11548) 2007-12-14 01:24 +0000 [r92875] Mark Michelson * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS, don't spam the CLI with messages about possible deadlocks. Instead just print the intended single message every five seconds. (closes issue 11537, reported and patched by dimas) 2007-12-13 21:28 +0000 [r92815] Tilghman Lesher * channels/chan_zap.c: Properly initialize polarity statuses, so that they are detected properly. Reported by: julianjm Patch by: julianjm (Closes issue #10238) 2007-12-13 20:13 +0000 [r92809] Jason Parker * main/pbx.c: Make application help text a little more clear about the use of extensions in a filename. 2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson * apps/app_voicemail.c: Prevent another potential fd leak * apps/app_voicemail.c: Prevent a possible fd leak. 2007-12-13 00:11 +0000 [r92696] Jason Parker * main/config.c, channels/chan_sip.c, channels/chan_h323.c, channels/chan_iax2.c: If a typo is found in a config file, we previous continued on with what was already loaded. We do not want to do this (see bug below for details). This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded. Isue #10690. 2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming * codecs/codec_zap.c: emit a warning message when we drop a G.729B CNG frame destined for the transcoder 2007-12-12 21:15 +0000 [r92617] Jason Parker * apps/app_meetme.c: Don't increment user count until after name has been recorded (if enabled). Issue 11048, tested by pep. 2007-12-12 19:40 +0000 [r92556] Russell Bryant * res/res_features.c: resolve compiler warning 2007-12-12 17:46 +0000 [r92510] Mark Michelson * res/res_features.c: Correctly detect where a dynamic feature was activated. Before this patch, the channel which initiated the bridge was always assumed to have been the one which activated the dynamic feature. This patch corrects this. (closes issue #11529, reported and patched by nic_bellamy) 2007-12-12 16:52 +0000 [r92463] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, configure.ac: Test directly for the API that fixed AST-2007-026, to ensure that older versions of PostgreSQL are no longer acceptable. (Closes issue #11526) 2007-12-12 16:08 +0000 [r92443] Mark Michelson * apps/app_queue.c: Removing an unused variable. 2007-12-11 19:51 +0000 [r92363] Joshua Colp * main/global_datastores.c: Fix potential memory leak with the dialed interfaces list if another memory allocation fails. (closes issue #11507) Reported by: eliel Patches: global_datastores.c.patch uploaded by eliel (license 64) 2007-12-11 17:42 +0000 [r92323] Mark Michelson * apps/app_queue.c: Fixing autofill to be more accurate. Specifically, if calls ahead of the current caller were ringing members (but not yet bridged) there could be available members and waiting callers who would not get matched up. The member availability checker was correctly determining the number of available members in this scenario, but the queue itself did not parallelly reflect this status on the pending calls. This commit corrects the issue. (closes issue #11459, reported by equissoftware, patched by me) 2007-12-10 16:36 +0000 [r92204] Joshua Colp * main/rtp.c: Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much. (closes issue #11483) Reported by: revolution Patches: rtp.diff uploaded by revolution (license 346) 2007-12-10 16:29 +0000 [r92202] Mark Michelson * apps/app_queue.c: If there are no members in a queue, then the loop where the datastore for detecting duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means that when we try to free it, there's a crash. This stops that crash from occurring. (closes issue #11499, reported by slavon, patched by eliel) 2007-12-10 16:13 +0000 [r92200] Joshua Colp * channels/chan_sip.c: It is possible for nativeformats to contain more then one codec, so print out multiple ones. (closes issue #11366) Reported by: ovi 2007-12-10 14:04 +0000 [r92158] Olle Johansson * channels/chan_sip.c: Avoid reinvite race situations with two Asterisks trying to reinvite each other in 1.4 and trunk. This patch implements support for the 491 error code that Asterisk 1.4 generates on situations where we get an incoming INVITE and already has one in progress. Thanks to mavetju for reporting and to Raj Jain for an excellent explanation of the problem. Patch by myself. Tested with 8 Asterisk servers connected to each other in a training network. Closes issue #10481 2007-12-07 23:29 +0000 [r91890] Jason Parker * main/dsp.c: We need to make sure we free the input frame if we return a different frame in ast_dsp_process. Issue 11273, pointed out by dimas, with a patch by eliel. 2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming * codecs/codec_zap.c: even though Asterisk explicitly requests that endpoints using G.729 do *not* use Annex B (silence detection and comfort noise generation) some do anyway; the transcoder card interface does not currently work properly with CNG frames, so trim off the CNG before sending the data 2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant * main/utils.c: Make the lock protecting each thread's list of locks it currently holds recursive. I think that this will fix the situation where some people have said that "core show locks" locks up the CLI. (related to issue #11080) * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS code. The ast_mutex_init() function had the mutex attribute object marked as static. This means that multiple threads initializing locks at the same time could step on each other and end up with improperly initialized locks. (found when tracking down locking issues related to issue #11080) * include/asterisk/lock.h: I love fixing lock related errors in the lock debugging code. That's about as ironic as it gets in Asterisk programming land. Anyway, I spotted this bug while trying to track down why systems are locking up and acting weird in issue #11080. The mutex attribute object was marked as static in this function when it should not have been. * apps/app_dial.c: * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Remove the dialed variable as it isn't needed. * Restructure some code for clarity and coding guidelines stuff * apps/app_queue.c: * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Handle memory allocation failure. * Remove the dialed variable, as it wasn't actually needed. * Tweak some formatting to conform to coding guidelines. * main/autoservice.c: * Add a bit more of a verbose comment as to why a hangup frame needs to be queued up if autoservice gets a NULL return from ast_read(). * Make the process of queueing the hangup frame more efficient by putting the frame where it is going to end up and avoiding some locking and extra memory allocations and freeing. 2007-12-07 15:39 +0000 [r91737] Mark Michelson * main/autoservice.c: Hangups that happen during autoservice were not processed appropriately. This is because a hangup actually causes a NULL frame to be received, not a hangup frame. Queueing a hangup if we receive a NULL frame during autoservice corrects this problem (closes issue #11467, reported by jmls, patched by me) 2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant * apps/app_dial.c: Don't unlock the dialed_interfaces list until we're done messing with the iterator. * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels from Queue() and Dial() again. There was a slight flaw in the code to prevent call forwards from looping that caused this problem. (related to issue #11486) * apps/app_queue.c: Fix in an issue in the call forwarding handling code that was causing crashes on every call into a queue. I'm not entirely sure about the logic in this part of the code, so I want to look at it some more tomorrow. However, this makes it safe and keeps it from crashing. (closes issue #11486, reported by adamg, patched by me) 2007-12-07 00:52 +0000 [r91637] Tilghman Lesher * main/rtp.c: At the end of a call, when we're reporting, RTCP may already be partially torn down, so check for NULL dereference Reported by: blitzrage Patch by: tilghman (Closes issue #11450) 2007-12-06 20:25 +0000 [r91541] Mark Michelson * apps/app_voicemail.c: IMAP storage did not honor the maxmsg setting in voicemail.conf, and it also had the possibility of crashing if a user had more than 256 messages in their voicemail. This patch kills two birds with one stone by adding maxmsg support and also setting a hard limit on the number of messages at 255 so that the crashes cannot happen. (closes issue #11101, reported by Skavin, patched by me) 2007-12-06 19:11 +0000 [r91501] Russell Bryant * main/loader.c, include/asterisk/module.h: Add a new module flag to indicate that a build sum is present. Modules built against older Asterisk 1.4 headers will now load properly with just a warning indicating that they are old and may cause problems. (patch by paravoid) 2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp * main/udptl.c: Fix various in the udptl implementation. It could return empty modem frames, have an incorrect sequence number on packets, and display the wrong sequence number in the debug messages. (closes issue #11228) Reported by: Cache Patches: udptl-4.patch uploaded by dimas (license 88) * channels/chan_sip.c: Add support for accepting and sending T.38 in the initial INVITE. (closes issue #9402) Reported by: thdei 2007-12-06 12:54 +0000 [r91366] Olle Johansson * main/loader.c, include/asterisk/logger.h, main/logger.c: Make sure logger is reloaded at general reload in the cli. (Discovered during Asterisk training in Portugal) 2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson * apps/app_voicemail.c: Reverting extra stuff I didn't mean to commit * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial() did not properly handle the case where only a label was provided. This was due to the fact that the answering channel did not have an extension set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto on the answering channel since it is a wasteful call. The answering channel and the calling channel are both directed to the same extension and context, just different priorities, so we can just copy the values from the calling channel to the answering channel and increment the answering channel's priority. (closes issue #11382, reported by jon, patch by me with correction by jon) 2007-12-05 21:38 +0000 [r91237] Tilghman Lesher * sounds/Makefile: Upgrade to the latest version of extra sounds 2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant * main/threadstorage.c: Make the lock in the threadstorage debugging code untracked to avoid a deadlock on thread destruction. (closes issue #11207) Reported by: ys Patches: threadstorage.c.diff uploaded by ys (license 281) Also fixes an open bug report: (closes issue #11446) * main/utils.c: When DEBUG_THREADS is enabled, we only have the details about who is holding a lock that we are waiting on for a mutex, not rwlocks. This should fix the problem where people have reported "core show locks" crashing sometimes. * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were reported as happening on Mac systems. It turns out that the problem was the Mac version of the ast_atomic_fetchadd_int() function. The Mac atomic add function returns the _new_ value, while this function is supposed to return the old value. So, the crashes happened on unreferencing objects. If the reference count was decreased to 1, ao2_ref() thought that it had been decreased to zero, and called the destructor. However, there was still an outstanding reference around. (closes issue #11176) (closes issue #11289) * include/asterisk/file.h, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/compiler.h: Modify file.h to maintain API compatibility with earlier versions. If a recent compiler is being used, then a warning will show up for any modules still using the old name "private" instead of "_private". (patch suggested by paravoid) * main/pbx.c: Make some changes to some additions I made recently for doing channel autoservice when looking up extensions. This code was added to handle the case where a dialplan switch was in use that could block for a long time. However, the way that I added it, it did this for all extension lookups. However, lookups in the in-memory tree of extensions should _not_ take long enough to matter. So, move the autoservice stuff to be only around executing a switch. 2007-12-04 17:28 +0000 [r90876] Jason Parker * main/channel.c: If we fail to create a channel after allocating a timing fd, we need to make sure to close it. Issue 11454, patch by eliel. 2007-12-04 05:29 +0000 [r90798] Joshua Colp * apps/app_dial.c: Fix build issue on the build cluster. 2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher * include/asterisk/compat.h: Solaris requires the inclusion of sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11430) * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a NULL deref could result Reported by: xrg Patch by: tilghman (Closes issue #11387) 2007-12-03 23:12 +0000 [r90735] Mark Michelson * apps/app_dial.c, main/channel.c, main/global_datastores.c (added), channels/chan_local.c, main/Makefile, include/asterisk/channel.h, include/asterisk/global_datastores.h (added), apps/app_queue.c: A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. 2007-12-03 22:06 +0000 [r90696] Jason Parker * apps/app_meetme.c: Make sure we always close the conference fd if we have an open one. Issue 11383, reported by markmhy, patch by eliel. 2007-12-03 20:59 +0000 [r90639] Mark Michelson * channels/chan_mgcp.c: Changing some bad logic when calculating the interdigit timeout. (closes issue #11402, reported and patched by eferro) 2007-12-03 20:51 +0000 [r90607] Jason Parker * res/res_features.c: Fix crash in ParkAndAnnounce application. Issue #11436, reported by lytledd, patch by eliel. 2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp * main/rtp.c: Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size. * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88) 2007-12-03 18:20 +0000 [r90546] Jason Parker * channels/chan_iax2.c: Only log debug messages if debug is enabled. Closes issue #11416, patch by casper. 2007-12-02 18:18 +0000 [r90470] Russell Bryant * apps/app_queue.c: The other day when I went through making changes as a result of the ao2_link() change, I added some code to set pointers to NULL after they were unreferenced. This pointed out that in this place, the object was unreferenced before the code was done using it. So, move the unref down a little bit. (crash reported by jmls on IRC) 2007-12-02 09:34 +0000 [r90432] Tilghman Lesher * main/autoservice.c: Clarify the return value on autoservice. Specifically, if you started autoservice and autoservice was already on, it would erroneously return an error. Reported by: adiemus Patch by: dimas (Closes issue #11433) 2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h, apps/app_queue.c, channels/chan_iax2.c: Change the behavior of ao2_link(). Previously, in inherited a reference. Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. * include/asterisk/astobj2.h: Add some notes on the behavior of ao2_unlink() after a discussion with Tilghman 2007-11-30 14:43 +0000 [r90269] Joshua Colp * channels/chan_sip.c: Fix locking issues under one legged replaces scenarios. (closes issue #11420) Reported by: irroot Patches: chan_sip_oneleg.patch uploaded by irroot (license 52) 2007-11-30 00:16 +0000 [r90231] Mark Michelson * channels/chan_mgcp.c: Clear the DTMF buffer if the call times out. (closes issue #11418, reported and patched by eferro) 2007-11-29 Russell Bryant * Asterisk 1.4.15 released. 2007-11-29 19:48 +0000 [r90166] Tilghman Lesher * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure we use thread-safe escaping (Fixes AST-2007-026) 2007-11-29 19:38 +0000 [r90163] Mark Michelson * apps/app_queue.c: This patch handles the case where a queue member with a negative penalty is added via the manager. If a negative value is submitted for a member penalty, we set it to 0. (closes issue #11411, reported and patched by Laureano) 2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher * res/res_config_pgsql.c: Properly escape input buffers (Fixes AST-2007-025) * formats/format_g726.c, include/asterisk/file.h, formats/format_wav.c, formats/format_pcm.c, formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c: Use of "private" as a field name in a header file messes with C++ projects Reported by: chewbacca Patch by: casper (Closes issue #11401) * sounds/Makefile: Upgrade the core sounds release version 2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant * funcs/func_callerid.c: fix some formatting i accidentally changed * funcs/func_callerid.c, main/channel.c, include/asterisk/channel.h: This set of changes is to make some callerID handling thread-safe. The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. * include/asterisk/file.h, main/file.c: Merge a change from team/russell/chan_refcount ... This makes ast_stopstream() thread-safe. 2007-11-28 22:59 +0000 [r90101] Joshua Colp * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405) Reported by: eliel Patches: load_realtime.patch uploaded by eliel (license 64) 2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming * configs/users.conf.sample, main/manager.c: it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) 2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson * main/pbx.c: Removing some seemingly pointless code. This sets a channel variable for every priority executed in the dialplan if you have debug set to anything non-zero. This seems pointless due to the fact that these channel variables are not referenced anywhere else in the code and their names are esoteric enough that they would not be practical to reference in the dialplan. Plus the fact that this behavior isn't documented anywhere means that the change is not likely to cause any disruption. If anything, this may actually cause a slight performance increase if running with debug on. The motivating influence for this code change is the eventwhencalled option for queues. If set to vars, all channel variables will be output to the manager. These unnecessary channel variables make the output a lot more difficult to deal with. * apps/app_voicemail.c: Recording greetings when using IMAP storage was causing zero-length files to be stored. Since greetings are not retrieved from IMAP anyway, it is pointless to attempt storing them there. (closes issue #11359, reported by spditner, patched by me) 2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant * main/pbx.c, include/asterisk/pbx.h: - update documentation for some of the goto functions to note that they handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed * main/autoservice.c: Don't do frame processing if ast_read() returned NULL. * apps/app_queue.c: Instead of depending on the return value of ast_true(), explicitly set the eventwhencalled variable to 1. * main/pbx.c: Don't start/stop autoservice in pbx_extension_helper() unless a channel exists 2007-11-27 23:10 +0000 [r89837] Mark Michelson * apps/app_queue.c: Two changes with regards to the 'eventwhencalled' option of queues.conf 1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' did exactly the same thing. Thus the sign change of the ast_true call. 2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting in bizarre output for the channel variables. This patch remedies this. (related to issue #11385, however I'm not sure if this will actually be enough to close it) 2007-11-27 21:45 +0000 [r89790] Russell Bryant * main/autoservice.c, main/pbx.c: Merge changes from team/russell/autoservice_1.4 This set of changes fixes an issue that was reported to me on IRC yesterday. The user, d1mas, was using chan_zap for incoming calls and was having DTMF recognition issues in some situations. Specifically, he noticed that the problem occurred when using DISA or WaitExten. He also noticed that when using Read, the problem did not occur. His system also used DUNDi for dialplan lookups. So, he theorized that if the DUNDi lookups blocked for some period of time, that audio from the zap channel could get lost. If the audio got lost, then it wouldn't be run through the DTMF detector, and digits could get lost. He was correct, and the following set of changes fixes the problem. However, the changes go a little bit further than what was necessary to fix this exact problem. 1) I updated pbx_extension_helper() to autoservice the associated channel to handle cases where extension lookups may take a long time. This would normally be a dialplan switch that does some lookup over the network, such as the DUNDi or IAX2 switches. This ensures that even while a DUNDi lookup is blocking, the channel will be continuously serviced. 2) I made a change to the autoservice code. This is actually something that has bothered me for a long time. When a channel is in autoservice, _all_ frames get thrown away. However, some frames really shouldn't be thrown away. The most notable examples are signalling (CONTROL) frames, and DTMF. So, this patch queues up important frames while a channel is in autoservice. When autoservice is stopped on the channel, the queued up frames get stuck back on the channel so that they can get processed instead of thrown away. 3) I made another change to the autoservice code to handle the case where autoservice is started on channels recursively. Previously, you could call ast_autoservice_start() multiple times on a channel, and it would stop the first time ast_autoservice_stop() gets called. Now, it will ensure that autoservice doesn't actually stop until the final call to ast_autoservice_stop(). 2007-11-27 20:22 +0000 [r89727] Mark Michelson * res/res_config_pgsql.c: Changing some calls from free() to ast_free() since they were allocated with ast_calloc(). (closes issue #11390, reported and patched by Laureano) 2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming * main/app.c: on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL * main/app.c: generate a warning when an application option that requires an argument is ignored due to lack of an argument 2007-11-27 16:12 +0000 [r89634] Russell Bryant * configs/voicemail.conf.sample: Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. 2007-11-27 15:38 +0000 [r89631] Tilghman Lesher * funcs/func_env.c: Default result of STAT should be "0" not "". Reported via the -users mailing list, fixed by me. 2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue #11304) Reported by: pj 2007-11-27 06:24 +0000 [r89622] Steve Murphy * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample, include/asterisk/cdr.h: closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. 2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson * apps/app_playback.c: After issuing a "say load new", if a caller hangs up during the middle of playback of a number, app_playback will continue to try to play the remaining files. With this change, no more files will be played back upon hangup. (closes issue #11345, reported and patched by IgorG) * apps/app_playback.c: After issuing a "say load new" tons of warning messages are printed out to the CLI every time do_say in app_playback is called. Removing these warnings 2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp * main/dial.c: Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) * res/res_features.c: Add module counting removal for error conditions. (closes issue #11333) Reported by: Laureano Patches: res_features_v2.c.patch uploaded by Laureano (license 265) 2007-11-26 17:41 +0000 [r89594] Russell Bryant * main/pbx.c: Add channel locking to a function that needed to be doing it. This is just a little something I noticed while working on a completely unrelated issue. 2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp * pbx/pbx_config.c: Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) * apps/app_mixmonitor.c: Close the audio file before sending it to the post processing application. (closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) 2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming * main/app.c: when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it 2007-11-26 15:48 +0000 [r89580] Mark Michelson * apps/app_voicemail.c: Revert vmu->email back to an empty string if it was empty when imap_store_file was called. This prevents sending a duplicate e-mail. (closes issue #11204, reported by spditner, patched by me) 2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp * main/channel.c: If channel allocation fails because the alert pipe could not be created also free the scheduler context. (closes issue #11355) Reported by: eliel Patches: main.channel.c.patch uploaded by eliel (license 64) * apps/app_meetme.c: When unloading app_meetme destroy any auto created contexts created by SLA. (closes issue #11367) Reported by: eliel 2007-11-25 17:17 +0000 [r89559] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample, include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously attempted to use the ESCAPE clause to set the escape delimiter to a backslash. Unfortunately, this does not universally work on all databases, since on databases which natively use the backslash as a delimiter, the backslash itself needs to be delimited, but on other databases that have no delimiter, backslashing the backslash causes an error. So the only solution that I can come up with is to create an option in res_odbc that explicitly specifies whether or not backslash is a native delimiter. If it is, we use it natively; if not, we use the ESCAPE clause to make it one. Reported by: elguero Patch by: tilghman (Closes issue #11364) 2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher * res/res_adsi.c: Free some frames that would otherwise leak on error. Reported by: Laureano Patch by: Laureano,tilghman (Closes issue #11351) * apps/app_voicemail.c, main/app.c: Currently, zero-length voicemail messages cause a hangup in VoicemailMain. This change fixes the problem, with a multi-faceted approach. First, we do our best to avoid these messages from being created in the first place, and second, if that fails, we detect when the voicemail message is zero-length and avoid exiting at that point. Reported by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083) * main/manager.c: Up until this point, the XML output of the manager has been technically invalid, due to the repetition of certain parameters in a single event. This caused various issues for XML parsers, some of which refused to parse at all, given the invalidity of the rendered XML. So this commit fixes the XML output, ensuring that each entity parameter has a unique name, thus ensuring valid XML. Reported by: msetim Patch by: tilghman (Closes issue #10220) * res/res_config_odbc.c: Use ESCAPE clause for the first parameter, not just 2nd-Nth parameters. Reported by: apsaras Patch by: tilghman (Closes issue #11353) 2007-11-22 17:29 +0000 [r89527] Russell Bryant * configs/agents.conf.sample: mvanbaak pointed out a spelling error in this sample configuration file. While I was at it, I went ahead and tweaked it a little bit more. 2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson * apps/app_queue.c: Fix a small error I made in my previous commit * apps/app_queue.c: Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this message would always report that there were 0 members available, even though that may not be true. 2007-11-21 18:59 +0000 [r89491] Terry Wilson * res/res_features.c: If a channel gets masqueraded in the middle of a park, don't play the announcement to the masqueraded channel, and dial back to the original channel on timeout. 2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming * include/asterisk/module.h: re-doxygen some comments * main/loader.c, include/asterisk/module.h, build_tools/make_buildopts_h: bring back compile-option checking when loading modules, only this time use a string-based storage and comparison mechanism because it is easier to support on other platforms 2007-11-20 17:50 +0000 [r89457] Mark Michelson * main/pbx.c: According to comments in main/pbx.c, it is essential that if we are going to lock the conlock as well as the hints lock, it must be locked in that respective order. In order to prevent a potential deadlock, we need to lock the conlock prior to locking the hints lock in ast_hint_state_changed (see the call stack example on issue #11323 for how this can happen). (closes issue #11323, reported by eelcob, suggestion for patch by eelcob, patch by me) 2007-11-20 15:22 +0000 [r89450] Steve Murphy * doc/queues-with-callback-members.txt: closes issue #11324; break statements missing in switch cases. 2007-11-20 13:40 +0000 [r89445] Christian Richter * channels/chan_misdn.c: added RR patch from iroot #10908, thanks. 2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp * res/res_features.c: Print out the correct filename (features.conf) in the log message when parkpos options are incorrect. (closes issue #11295) Reported by: Laureano Patches: res_features.c.patch uploaded by Laureano (license 265) * doc/localchannel.txt: Clarify documentation a bit, include that a frame has to pass through the core in order for the Local channel optimization to happen. (closes issue #11246) Reported by: jon 2007-11-16 Russell Bryant * Asterisk 1.4.14 released. 2007-11-16 22:26 +0000 [r89339] Russell Bryant * main/loader.c, include/asterisk/module.h, build_tools/make_buildopts_h: Temporarily revert revision 89325, which added md5 magic for keeping track of what build options were used. We agreed that we should remove this before making a 1.4 release, and then we can put it back in. Then, we can take a month or so to play around with it to get it how we want it. 2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming * main/loader.c, include/asterisk/module.h, build_tools/make_buildopts_h: To help combat problems where people build external modules (asterisk-addons or others) and then change the build options of the Asterisk build in a way that makes the incompatible without warning, this commit introduces an MD5 signature of the important build-time options and includes that signature into modules when they are built. When the loader loads one of these modules and notices the problem, it will emit a warning to console and refuse to initialize the module, as doing so could cause the system to be unstable or even crash. If you upgrade to this version of Asterisk, you must rebuild *all* of your modules that came from other sources before trying to run this version. If you are using Digium's G.729 binary codec module, you will need v33 or newer. 2007-11-16 15:28 +0000 [r89323] Mark Michelson * apps/app_queue.c: Make realtime queues accessible from the QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and patched by atis, with small modifications from me) 2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher * Makefile: Start Asterisk in Debian at a more reasonable time (since zaptel is at level 20) * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found by valgrind * channels/chan_iax2.c: Yet another memory corruption issue. Reported by: atis Patch by: tilghman Fixes issue #10923 2007-11-15 17:19 +0000 [r89296] Russell Bryant * apps/app_meetme.c: Update the SLAStation application to account for the case where the SLA thread has a call out to the station, but the user has pressed a line button to answer the call instead of picking up the handset. If they do, the phone sends out a new INVITE. So, the SLAStation app must check to see if it is picking up a ringing trunk, and ensure that the other stations stop ringing. (reported internally, patched by me, tested by mogorman) 2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson * main/manager.c: Undoing previous commit since I realize it was wrong * main/manager.c: Adding a missing mutex unlock. (closes issue 11256, reported and patched by ys) 2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson * channels/chan_sip.c: Don't send re-invites during pending INVITE transactions. Patch by one47 - thanks! Closes issue #9305 * channels/chan_sip.c: Improve support for multipart messages. Code by gasparz, changes by me (mostly formatting). Thanks, gasparz! Closes issue #10947 2007-11-14 23:23 +0000 [r89275] Tilghman Lesher * main/app.c: When a recording ends with '#', we are improperly trimming an extra 200ms from the recording. Reported by: sim Patch by: tilghman Closes issue #11247 2007-11-14 01:15 +0000 [r89260] Joshua Colp * main/srv.c: Return the proper value when the srv_callback function executes properly. (closes issue #11240) Reported by: jtodd 2007-11-13 21:07 +0000 [r89248-89254] Jason Parker * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer systems which require a third arg to open() when using O_CREAT. Issue 11238, reported by puzzled. * res/res_features.c: Revert change from revision 67064. It is documented behavior that if a parking extension already exists while using PARKINGEXTEN, dialplan execution will continue. If blind transferring to a Park with PARKINGEXTEN, you must keep this in mind, and handle the failure yourself. Issue 11237, reported by jon. 2007-11-13 17:34 +0000 [r89246] Tilghman Lesher * channels/chan_sip.c: If we set a value for qualify, we should actually pay attention to it, instead of overriding the value 2007-11-13 16:02 +0000 [r89241] Mark Michelson * apps/app_mixmonitor.c: Reverting commit made in revision 89205 since it is unnecessary. Thanks to Kevin for pointing this out 2007-11-13 13:51 +0000 [r89239] Tilghman Lesher * main/utils.c: Debugging is running into the 16-lock limit. Increase to avoid. (This define is only effective when debugging is turned on, so there's no effect for most installations.) 2007-11-13 00:56 +0000 [r89205] Mark Michelson * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options and args.post_process strings are uninitialized and could contain garbage. This change handles this situation properly by only using arguments that we have parsed. 2007-11-12 20:46 +0000 [r89194] Jason Parker * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev 2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher * main/config.c: If two config writes collide, file corruption could result. Use a mkstemp() file, instead. Reported by: paravoid Patch by: tilghman Closes issue #10781 * main/channel.c, channels/chan_sip.c: Fix two cases of memory corruption caused by background threads. Reported by: atis Patch by: tilghman Fixes issue #10923 2007-11-12 11:26 +0000 [r89169-89173] Christian Richter * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) * channels/misdn/isdn_lib.c: fixed the support for CW and therefore for the reject_cause option. * 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: aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. 2007-11-08 23:52 +0000 [r89125] Jason Parker * main/say.c: Properly say the seconds here.. Issue 11203, fix described by vma. 2007-11-08 21:00 +0000 [r89119] Mark Michelson * channels/chan_sip.c: Rework of the commit I made yesterday to use the already built-in ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction 2007-11-08 18:45 +0000 [r89115] Jason Parker * configs/res_odbc.conf.sample: Avoid warnings on load when using sample configuration files. Issue 11195, patch by eliel. 2007-11-08 16:47 +0000 [r89111] Mark Michelson * apps/app_voicemail.c: I made this same adjustment in trunk to fix a bug, and it makes sense to do it in 1.4 as well. If an imapfolder is specified in voicemail.conf, don't ever explicitly connect to INBOX since it may not exist. 2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming * main/srv.c: fix a glaring bug in the new SRV record handling that would cause incorrect weight sorting 2007-11-08 04:55 +0000 [r89103] Tilghman Lesher * doc/valgrind.txt: Typo 2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp * channels/chan_sip.c: Do not add a sip: to the beginning of the To URI unless needed. (closes issue #10756) Reported by: goestelecom * channels/chan_sip.c: Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) * channels/chan_sip.c: Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres * channels/chan_sip.c: If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. (closes issue #11185) Reported by: spditner 2007-11-07 23:39 +0000 [r89093] Tilghman Lesher * apps/app_queue.c: The member refcount must be incremented, to avoid using it after deallocation. A huge thanks go to lvl- for patiently providing the necessary valgrind output that was necessary to finding this problem of memory corruption. Reported by: lvl- Patch by: tilghman Closes issue #11174 2007-11-07 22:40 +0000 [r89090] Mark Michelson * channels/chan_sip.c: This patch makes it possible for SIP phones to dial extensions defined with '#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) 2007-11-07 21:40 +0000 [r89088] Steve Murphy * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho. 2007-11-07 15:56 +0000 [r89085] Mark Michelson * main/manager.c: Fixing a segfault in the manager "core show channels concise" command. (closes issue #11183, reported by arnd and patched by ys) 2007-11-07 04:07 +0000 [r89079] Tilghman Lesher * configs/extensions.ael.sample: Suppress AEL warnings on load. Reported by: eliel Patch by: eliel Closes issue #11178 2007-11-06 20:18 +0000 [r89053] Russell Bryant * res/res_musiconhold.c: Fix init_classes() so that classes that actually do have files loaded aren't treated as empty, and immediately destroyed ... 2007-11-06 19:09 +0000 [r89046] Jason Parker * codecs/codec_zap.c: Correctly set the total number of channels from a zaptel transcoder board. SPD-49, patch by Matthew Nicholson. 2007-11-06 19:09 +0000 [r89045] Tilghman Lesher * include/asterisk/lock.h: We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops). 2007-11-06 18:53 +0000 [r89042] Olle Johansson * main/tdd.c: Bug fixes to tdd support in zaptel. 2007-11-06 18:20 +0000 [r89037] Russell Bryant * res/res_musiconhold.c: If someone were to delete the files used by an existing MOH class, and then issue a reload, further use of that class could result in a crash due to dividing by zero. This set of changes fixes up some places to prevent this from happening. (closes issue #10948) Reported by: jcomellas Patches: res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282) Additional changes added by me. 2007-11-06 17:52 +0000 [r89036] Steve Murphy * main/config.c: closes issue #8786 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix. 2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp * channels/chan_sip.c: Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue #11006) Reported by: pj * channels/chan_zap.c: Fix improbable but possible memory leaks in chan_zap. (closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) 2007-11-06 13:50 +0000 [r88931] Russell Bryant * include/asterisk/lock.h: Remove some checks to see if locks are initialized from the non-DEBUG_THREADS versions of the lock routines. These are incorrect for a number of reasons: - It breaks the build on mac. - If there is a problem with locks not getting initialized, then the proper fix is to find that place and fix the code so that it does get initialized. - If additional debug code is needed to help find the problem areas, then this type of things should _only_ be put in the DEBUG_THREADS wrappers. 2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming * include/asterisk/srv.h: update comment to match the state of the code 2007-11-05 23:29 +0000 [r88826] Mark Michelson * main/channel.c: Reworked deadlock avoidance in __ast_read. Restored audio to callback agents. (closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown) 2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related to channel variables, I went looking around at the ways that channel variables are handled. In general, they were not handled in a thread-safe way. The channel _must_ be locked when reading or writing from/to the channel variable list. What I have done to improve this situation is to make pbx_builtin_setvar_helper() and friends lock the channel when doing their thing. Asterisk API calls almost all lock the channel for you as necessary, but this family of functions did not. (closes issue #10923, reported by atis) (closes issue #11159, reported by 850t) * channels/chan_sip.c: When traversing the list of channel variables here in transmit_invite(), the asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) * channels/chan_sip.c: Fix up some indentation. * main/srv.c, include/asterisk/srv.h: Merge changes from asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV record support in Asterisk was broken. There was no guarantee on what record Asterisk would choose to actually use. This set of changes improves the situation by ensuring that Asterisk will choose the highest priority record. * main/channel.c: Merge the last bit of changes from asterisk/team/russell/readq-1.4 The issue here is that the channel frame readq handling got broken when the code was converted to use the linked list macros. It caused corruption of the list head and tail pointers. So, I fixed up the usage of the linked list macros and in passing, simplified the code. I also documented what the code is doing, as it was a bit difficult to figure out at first. This bug showed itself with crashes showing messed up head/tail pointers for the readq. However, there are a couple of crashes that aren't quite as obvious, but I think may be related. So, if your bug gets closed by this commit, but you still have a problem, please reopen or create a new bug report. (closes issue #10936) (closes issue #10595) (closes issue #10368) (closes issue #11084) (closes issue #10040) (closes issue #10840) 2007-11-05 18:47 +0000 [r88671] Joshua Colp * channels/chan_sip.c: If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue #11085) Reported by: francesco_r Tested by: blitzrage (closes issue #10474) Reported by: acennami 2007-11-05 17:46 +0000 [r88624] Russell Bryant * main/channel.c: Fix up datastore handling in ast_do_masquerade(). The code is intended to move any channel datastores from the old channel to the new one. However, it did not use the linked list macros properly to accomplish the task. The existing code would only work if there was only a single datastore on the old channel. 2007-11-05 17:19 +0000 [r88585] Jason Parker * channels/chan_sip.c: Make sure we destroy the config structure on configuration failure. Issue 11163, patch by eliel. 2007-11-05 16:20 +0000 [r88539] Tilghman Lesher * res/res_odbc.c: Don't check used pooled connections for connection status, as it will cause issues for prepared queries. Reported by: Nick Gorham (via -dev list) Patch by: tilghman 2007-11-04 22:38 +0000 [r88471] Luigi Rizzo * include/asterisk/stringfields.h, main/channel.c, apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c: Rename ast_string_field_free_pool to ast_string_field_free_memory, and ast_string_field_free_all to ast_string_field_reset_all to avoid misuse (due to too similar names and an error in documentation). Fix two related memory leaks in app_meetme. No need to merge to trunk, different fix already applied there. Not applicable to 1.2 2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp * channels/chan_sip.c: Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context). (closes issue #10702) Reported by: slavon * channels/chan_sip.c: If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. (closes issue #5747) Reported by: chandi Patches: infofix-81430-1.patch uploaded by IgorG (license 20) 2007-11-02 16:51 +0000 [r88283] Jason Parker * main/say.c: We need to make sure to specify a language to ast_fileexists, otherwise it may fail for anything besides en Issue 11147, fix discovered by both citats and myself (independently), with input from Corydon76 2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy Patch by: ys Closes issue #11143 * doc/valgrind.txt (added): Add some notes on using valgrind 2007-11-01 16:21 +0000 [r88078] Jason Parker * channels/chan_zap.c: Make sure we set the poll fds to NULL after free()ing it. Part of issue 11017, patch by tzafrir. 2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp * apps/app_meetme.c: Fix up commit for my Zap channel with spies in Meetme fix. (thanks Tony Mountifield!) * apps/app_meetme.c: If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides. (closes issue #10060) Reported by: mparker 2007-10-31 21:23 +0000 [r87906-87908] Jason Parker * res/res_jabber.c: Make sure we free some allocated memory before returning. Issue 11131, patch by eliel. * channels/chan_gtalk.c: Don't try to allocate memory that we're just going to re-allocate later anyways. Issue 11130, patch by eliel. 2007-10-31 18:03 +0000 [r87852] Tilghman Lesher * Makefile: Create samples for ALL of the available options in asterisk.conf 2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan save' cli command saves a file where the semicolon is not escaped. Fixed this; User also wanted comments to be preserved across dialplan save, but this is impossible at this point in time, because comments are not stored in the dialplan. They are 'compiled' out of extensions.conf. The only way to preserve those comments is to use the config file reader/writer that the GUI uses to allow online user edits. extensions.conf is first and foremost, a config file, and is read in by the normal config-file reading routines. Then, it is processed into a dialplan (context/exten structs). * pbx/pbx_ael.c: Included some verbage in the check_includes func, to inform the user that included contexts that have no match in the AEL, might be OK, as AEL cannot check in the extensions.conf or the in-memory contexts, as they may not be there at the time of the check. 2007-10-30 23:02 +0000 [r87739] Tilghman Lesher * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD Reported by: ys Fixed by: ys Closes issue #11116 2007-10-30 21:19 +0000 [r87686] Russell Bryant * channels/chan_iax2.c: Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a race condition related to the handling of POKEing peers. Essentially, a reference to a peer is held by the scheduler when there are pending callbacks, but the reference count didn't reflect it. So, it was possible for a peer to hit a reference count of zero and have its destructor begin to be called at the same time that the scheduler thread ran a POKE related callback. If that happened, a crash would likely occur. (closes issue #11082, closes issue #11094) 2007-10-30 20:29 +0000 [r87650] Jason Parker * channels/Makefile: Only try to clean out h323/ if the h323/Makefile exists. 2007-10-30 16:13 +0000 [r87571] Joshua Colp * res/res_features.c: Add two more checks before printing out a warning message about bridging. If either channel has hungup of course the bridge will have failed. (closes issue #10009) Reported by: dimas 2007-10-30 15:45 +0000 [r87567] Jason Parker * main/editline/np/vis.c: Fix build of editline on Solaris. Issue 11113, patch by snuffy. 2007-10-30 15:10 +0000 [r87534] Joshua Colp * apps/app_followme.c: Return 1.4 to a state where it builds. Changing the arguments to a function and not changing where they are used is bad, mmmk? 2007-10-30 14:31 +0000 [r87514] BJ Weschke * apps/app_followme.c: Fix issue where the recorded name wasn't getting removed correctly. (closes issue #11115) Reported by: davevg Patches: followme-v3.diff 2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming * codecs/gsm: missed one directory * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs, codecs/lpc10, main/db1-ast, main/editline, main, codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile, codecs/lpc10/Makefile, utils, codecs, agi, main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr: clean up (and ignore) assembler and preprocessor intermediate files if any are created during the build * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is already there (used when debugging preprocessor issues) because the compiler will whine about each compile command 2007-10-29 21:06 +0000 [r87427] Mark Michelson * apps/app_voicemail.c: Removing a completely unnecessary quota check from IMAP code. 2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant * main/utils.c, include/asterisk/lock.h: Add some more details to the output of "core show locks". When a thread is waiting for a lock, this will now show the details about who currently has it locked. (inspired by issue #11100) * main/astmm.c: Remove a lock that doesn't make any sense. The regions lock needs to be held when traversing the list of allocated chunks so that they can be printed out to the CLI. (Thanks to eliel on #asterisk-dev for pointing this out!) 2007-10-29 17:20 +0000 [r87342] Joshua Colp * channels/chan_sip.c: Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason. (closes issue #9566) Reported by: atca_pres Patches: bug9566.patch uploaded by oej 2007-10-29 17:13 +0000 [r87340] Jason Parker * funcs/func_realtime.c, funcs/func_cut.c: Allow some function modules to compile under dev mode. Issue 11104, patch by andrew. 2007-10-29 14:23 +0000 [r87294] Joshua Colp * main/utils.c: Fix issue with ast_unescape_semicolon going into an endless loop. (closes issue #10550) Reported by: ramonpeek Patches: unescape-85177-1.patch uploaded by IgorG (license 20) 2007-10-28 13:46 +0000 [r87262] Tilghman Lesher * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c: Add autoservice to several more functions which might delay in their responses. Also, make sure that func_odbc functions have a channel on which to set variables. Reported by russell Fixed by tilghman Closes issue #11099 2007-10-26 16:34 +0000 [r87168] Steve Murphy * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h, pbx/ael/ael.tab.h, utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael.flex: closes issue #11086 where a user complains that references to following contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config, 2007-10-26 13:54 +0000 [r87120] Tilghman Lesher * funcs/func_curl.c: The addition of autoservice to func_curl additionally made func_curl dependent on the existence of a channel, with no real reason. This should make func_curl once again work without a channel. Reported by jmls. Fixed by tilghman. Closes issue #11090 2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming * main/channel.c, include/asterisk/linkedlists.h: appending one list to another should leave the first list empty, and not require the user to do that 2007-10-25 22:53 +0000 [r87067] Tilghman Lesher * funcs/func_cut.c: Backport alternate encoding of newline delimiters from trunk to 1.4, as approved by Russell Reported by blitzrage Closes issue #10903 2007-10-24 20:56 +0000 [r86982] Jason Parker * channels/chan_zap.c: Correctly respect hidecalleridname configuration option. Simplify code slightly in the process. Issue 11079, reported by ddv2005 2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable to specify app:spec in hint arguments * funcs/func_logic.c: closes issue #11052 -- where nothing after the ? will allow un-initialized variable values to corrupt and crash asterisk on 64-bit platforms * main/Makefile: this update to Makefile corrects how ast_expr2f.c should be generated * main/ast_expr2f.c: This should get rid of a really, really irritating warning generated by some 64-bit platforms from libc, where free(0) is frowned upon 2007-10-22 21:36 +0000 [r86836] Russell Bryant * include/asterisk/lock.h: If lock tracking is not enabled, then we can not attempt to log any mutex failures. If so, we could end up in infinite recursion. The only lock that is affected by this is a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys (license 281) 2007-10-22 17:38 +0000 [r86787] Tilghman Lesher * main/astmm.c: Minor FreeBSD build fix 2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp * channels/chan_sip.c: After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well. (closes issue #10113) Reported by: makoto * apps/app_controlplayback.c: Make sure res is a positive value before performing the check to determine whether the user stopped it or not. (closes issue #11023) Reported by: cfc 2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant * main/channel.c: Don't leak a frame in the case that an END frame is received and the time since the BEGIN is less than that of the defined minimum DTMF duration. (closes issue #11051) Reported by: casper Patches: channel.c.86664.diff uploaded by casper (license 55) * include/asterisk/lock.h: Update the static mutex initializer to include the initialization of the internal mutex used to protect the lock debugging data. (closes issue #11044, patch suggested by Ivan) 2007-10-22 14:48 +0000 [r86694] Mark Michelson * apps/app_voicemail.c: Account for the fact that sometimes headers may be terminated with \r\n instead of just \n (closes issue #11043, reported by yehavi) 2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp * main/channel.c: Move log message to before the frame it references is freed. (closes issue #11050) Reported by: slavon Patches: channel.c.86662.diff uploaded by casper (license 55) * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes issue #11041) Reported by: jsmith Patches: asterisk-dundicomplete.diff.txt uploaded by jamesgolovich (license 176) * main/loader.c: Fixes for building under OpenSolaris. (closes issue #11047) Reported by: snuffy Patches: 11047-fixes.diff uploaded by snuffy (license 35) 2007-10-22 09:21 +0000 [r86598] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan does not match after an overlap call. Also added out_cause=1 2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp * main/app.c: When returning a DTMF digit from ast_control_streamfile cast it as a char so that 0 does not overlap with the success return code. (closes issue #11023) Reported by: cfc * channels/chan_sip.c: Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote. (closes issue #11027) Reported by: ramonpeek Patches: 11027-1.diff uploaded by ramonpeek (license 266) * channels/chan_sip.c: Set port number in received as information for registrations as well. (closes issue #11028) Reported by: brad-x 2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development * apps/app_osplookup.c: Fixed OSP module did not report source/devinfo IP in correct format. 2007-10-18 22:01 +0000 [r86405-86406] Jason Parker * Makefile: Correct documentation. I removed the wrong line.. * Makefile: Add documentation for options in asterisk.conf Issue 11029, patch by eserra 2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous commit. * configs/iax.conf.sample, channels/chan_iax2.c: Add support for setting the maximum trunk size for IAX2 trunking * main/channel.c, include/asterisk/channel.h: The channel needs to stay locked while running timer callbacks, as they access and modify channel data that may change elsewhere. I went through every timer callback in the source tree to make sure that none of them did any additional locking that could introduce deadlocks, and all is well. (closes issue #10765) Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229) 2007-10-18 17:38 +0000 [r86328] Mark Michelson * apps/app_queue.c: If a non-existent file is specified to be played either as a periodic announcement or as a hold/position announcement, the caller would be kicked out of the queue. No longer does this happen. 2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant * codecs/codec_zap.c: Execute the RELEASE operation on transcoder channels in the destroy callback. (patch from jsloan) * main/utils.c: Revert a change that I made for issue #10979 which, as has been pointed out to me in issue #11018, doesn't really make sense. There is no reason to have the base64 decode function force a '\0' terminated buffer, when the result is almost always binary, anyway. In fact, this caused some breakage, as some code in res_crypto passed in a buffer exactly the right size to get its binary result, which got stomped on by this patch. (closes issue #11018, reported by dimas) 2007-10-17 21:39 +0000 [r86202] Mark Michelson * apps/app_queue.c: Changing the strategy field of the call_queue struct to be signed instead of unsigned, since the code attempts to set the strategy to -1 if you specify a bogus strategy. While this isn't a huge issue in 1.4, it could be a problem for someone who, say, tries to use the roundrobin strategy in trunk (despite all the deprecation warnings in 1.4). 2007-10-17 17:57 +0000 [r86149] Russell Bryant * channels/chan_sip.c: If Asterisk is in the middle of shutting down, respond to OPTIONS with 503 Unavailable. (closes issue #10994) Reported by: eserra Patches: sip-options-503.patch uploaded by eserra (license 45) 2007-10-17 16:58 +0000 [r86117] Joshua Colp * channels/chan_sip.c: Whoops, forgot to remove the original sip_scheddestroy. (closes issue #11010) Reported by: vadim 2007-10-17 15:23 +0000 [r86066] Tilghman Lesher * main/asterisk.c: When runuser/rungroup is specified, a remote console could only be attained by root (Closes issue #9999) 2007-10-17 15:06 +0000 [r86063] Joshua Colp * channels/chan_sip.c: Don't schedule dialog destruction if a MESSAGE is received using an existing dialog. (closes issue #11010) Reported by: vadim 2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson * configs/queues.conf.sample: Since monitor-join is deprecated now, remove the example from the sample queues.conf file * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of the monitor-join queue option * apps/app_queue.c: Adding deprecated warning to monitor-join option, since the plan is to no longer support this in favor of monitor-type = mixmonitor (related to issue #10885) 2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant * include/asterisk/lock.h: really picky formatting tweak ... * include/asterisk/lock.h: Some locking errors exposed the fact that the lock debugging code itself was not thread safe. How ironic! Anyway, these changes ensure that the code that is accessing the lock debugging data is thread-safe. Many thanks to Ivan for finding and fixing the core issue here, and also thanks to those that tested the patch and provided test results. (closes issue #10571) (closes issue #10886) (closes issue #10875) (might close some others, as well ...) Patches: (from issue #10571) ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license 229) - a few small changes by me 2007-10-16 21:14 +0000 [r85958] Mark Michelson * apps/app_queue.c: Trying to remove a non-dynamic queue member via dynamic means can lead to some interesting (read nasty) situations. This patch clears up the issue by making only dynamic queue members removable via dynamic methods. 2007-10-16 19:41 +0000 [r85921] Tilghman Lesher * main/stdtime/localtime.c: Also set up gmtoff (this is used in the %z gnu extension to strftime) Reported and fixed by jcmoore Closes issue #11002 2007-10-16 19:10 +0000 [r85896] Russell Bryant * apps/app_voicemail.c: Remove a pointless lock. 2007-10-16 15:21 +0000 [r85852] Mark Michelson * apps/app_queue.c: Fixing a double free which happens in the statechange thread. (closes issue #10987, reported by andrew) 2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp * apps/app_hasnewvoicemail.c: Check to make sure a value has been given to the VMCOUNT dialplan function. (closes issue #10996) Reported by: marsosa * main/threadstorage.c: Fix memory allocation issue in threadstorage. (closes issue #10995) Reported by: snuffy Patches: new-patch.diff uploaded by snuffy (license 35) 2007-10-16 10:46 +0000 [r85800] Philippe Sultan * channels/chan_gtalk.c: Fix the output for this channel help CLI command 2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant * apps/app_queue.c: Ensure that no pending state changes are leaked when the device state change thread gets stopped on module unload. * apps/app_queue.c: Previously, app_queue created a thread to handle every single device state change. I changed this a while ago in trunk for performance reasons. However, bug 8407 points out that it is actually a race condition, causing device state changes to get processed in random order. So, I backported my changes from trunk to 1.4. (closes issue #8407, patch provided by tim_ringenbach, committed patch by me) 2007-10-15 20:29 +0000 [r85687] Tilghman Lesher * apps/app_stack.c: Don't execute a gosub if the arguments is zero-len (not just NULL) Reported by davevg Fixed by me Closes issue #10985 2007-10-15 20:21 +0000 [r85686] Russell Bryant * main/say.c: Add a small fix for the tw version of saying dates. (closes issue #7827) Reported by: sharkey Patches: say.nits.patch uploaded by sharkey (license 172) 2007-10-15 20:15 +0000 [r85684] Jason Parker * Makefile: Properly use DESTDIR in 'config' target. Do not try to run chkconfig or similar if using DESTDIR. Issue 10938, patch by cabal95. 2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant * main/utils.c: Be pedantic about handling memory allocation failure. * main/utils.c: The loop in the handler for the "core show locks" could potentially block for some amount of time. Be a little bit more careful and prepare all of the output in an intermediary buffer while holding a global resource. Then, after releasing it, send the output to ast_cli(). * channels/chan_sip.c: Make the default for the srvlookup option to be yes. It doesn't really make sense for it to default to off. The default configuration file has it on, and proper RFC behavior, as indicated by a comment in the code, is for it to be on. So, let's have it on by default to make lives easier. (closes issue #10954, suggested by jtodd) 2007-10-15 16:39 +0000 [r85571] Joshua Colp * configs/features.conf.sample: Document that DTMF based features only work when two channels are bridged together. (closes issue #10773) Reported by: pbayley 2007-10-15 16:34 +0000 [r85561] Russell Bryant * include/asterisk/strings.h: Make a few changes so that characters in the upper half of the ISO-8859-1 character set don't get stripped when reading configuration. (closes issue #10982, dandre) 2007-10-15 16:22 +0000 [r85559] Joshua Colp * main/rtp.c: Bring both DTMF begin and end frames up through to the core for DTMF feature handling. (closes issue #10826) Reported by: dimas 2007-10-15 15:40 +0000 [r85556] Russell Bryant * pbx/pbx_dundi.c: Ensure the buffer passed to ast_canmatch_extension() is properly initialized so that it is null terminated. (issue #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by dimas (license 88) - small mods by me 2007-10-15 14:55 +0000 [r85552] Joshua Colp * main/rtp.c: If Monitor or a spy was added to a P2P or native bridged channel bring the channel back to the generic bridging core so the monitor or spy operations work. (closes issue #10943) Reported by: julianjm 2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled. (closes issue #10980) Reported by: casper Patches: db.c.84633.diff uploaded by casper (license 55) * main/asterisk.c: Make sure remote consoles unmute themselves again after reconnecting. (closes issue #10847) Reported by: atis Patches: console_unmute_on_reconnect.patch uploaded by atis (license 242) * main/utils.c: Make sure that the base64 decoder returns a terminated string. (closes issue #10979) Reported by: ys Patches: util.c.diff uploaded by ys (license 281) - small mods by me * pbx/pbx_config.c: Don't create the context for users in users.conf until we know at least one user exists. (closes issue #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by dimas (license 88) 2007-10-13 15:26 +0000 [r85536] Tilghman Lesher * configs/extensions.ael.sample: Remove deprecated syntax from sample ael file Reported and patched by: dimas Closes issue #10967 2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an issue with console verbosity when running asterisk -rx to execute a command and retrieve its output. The issue was that there was no way for the main Asterisk process to know that the remote console was connecting in the -rx mode. The way that James has fixed this is to have all remote consoles muted by default. Then, regular remote consoles automatically execute a CLI command to unmute themselves when they first start up. (closes issue #10847) Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded by jamesgolovich (license 176) * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly handle the case where read() may return the text for more than one CLI command at once for a remote console. (closes issue #10888) Reported by: jamesgolovich Patches: asterisk-climultiple.diff.txt uploaded by jamesgolovich (license 176) 2007-10-12 18:30 +0000 [r85523] Tilghman Lesher * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address 2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not SDMI. (closes issue #10959) * pbx/pbx_realtime.c: Fix the potential use of an uninitialized buffer in a log message. (closes issue #10958) Reported by: dimas Patches: realtime.patch uploaded by dimas (license 88) 2007-10-11 15:26 +0000 [r85397] Joshua Colp * channels/chan_sip.c: When creating a new packet don't try to stop retransmission of it. It was just allocated/created so it's impossible for it to have already been scheduled. (closes issue #10945) Reported by: flefoll Patches: chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll (license 244) 2007-10-11 04:35 +0000 [r85356] Tilghman Lesher * main/pbx.c: A dollar sign by itself, not indicating a start of a variable or expression prematurely ends substitution (closes issue #10939) 2007-10-10 Russell Bryant * Asterisk 1.4.13 released. 2007-10-10 15:56 +0000 [r85316] Russell Bryant * include/asterisk/file.h: I introduced a new member to the ast_filestream struct in 1.4.12, but put it in the middle of the struct, instead of at the end. One of the Debian folks, paravoid, pointed out that this breaks binary compatability with modules compiled against older headers. So, I'm moving the new member to the end of the struct to resolve the situation. 2007-10-10 15:51 +0000 [r85315] Mark Michelson * main/utils.c: The thread ID should be unsigned. 2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp * channels/chan_sip.c: If devicestate is passed a port number strip it out. (closes issue #10930) Reported by: ibc * channels/chan_sip.c: Add support for handling a 182 Queued response. (closes issue #10924) Reported by: ramonpeek Patches: queued-182.diff uploaded by ramonpeek (license 266) 2007-10-10 14:26 +0000 [r85276] Mark Michelson * apps/app_voicemail.c: A bunch of changes from sprintf to snprintf. See security advisory AST-2002-022 2007-10-10 14:14 +0000 [r85242] Joshua Colp * apps/app_voicemail.c: Close voicemail message description file if duration did not meet the minimum, or else we will eventually run out of file descriptors. (closes issue #10918) Reported by: brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license 279) 2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming * include/asterisk/frame.h: use a macro instead of an inline function, so that backtraces will report the caller of ast_frame_free() properly 2007-10-09 21:55 +0000 [r85158] Tilghman Lesher * main/channel.c, main/utils.c, include/asterisk/lock.h: This commit fixes the following issues: - Deadlock in ast_write (issue #10406) - Deadlock in ast_read (issue #10406) - Possible mutex initialization error in lock.h (issue #10571) 2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp * channels/chan_sip.c: Don't perform a reinvite if a transfer is in progress. (issue #10915) Reported by: ramonpeek * main/rtp.c: Only update codec information if the channel has a technology private structure. (issue #10915) Reported by: ramonpeek * main/rtp.c: Update codec information as well as address when doing hold reinvites. (issue #10868) Reported by: mavince * main/channel.c: Don't keep trying to native bridge if either of the channels are involved in a masquerade operation to be done. (closes issue #10696) Reported by: tbelder 2007-10-08 03:28 +0000 [r84957] Russell Bryant * Makefile.rules: Enable file dependency tracking for _all_ builds, and not just for builds with dev-mode enabled. I have seen enough problems caused by this that I don't think it's worth keeping. I want to continue to encourage anybody that is interested to continue to run Asterisk from svn. Furthermore, I do not want their systems to break when we change a structure definition in a header file. :) 2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan * res/res_jabber.c: Presence packets from a client who's connected with our Jabber ID are valid, therefore, those clients must be considered as buddies. The resource string helps us make the distinction between clients. Closes issue #10707, reported by yusufmotiwala. * res/res_jabber.c: Prevent Asterisk from crashing when receiving a presence packet without resource from a buddy that is known to have a resource list. Revert a change I previously made, where Asterisk could point to a freed memory location. 2007-10-05 19:42 +0000 [r84851] Tilghman Lesher * main/db.c: Log exactly why we can't open the database, if we fail (closes issue #10887) 2007-10-05 18:55 +0000 [r84818] Joshua Colp * main/rtp.c: Update the remembered RTP peer information when putting an endpoint on hold or taking it off hold so that the RTP stack does not initiate a needless reinvite. (closes issue #10868) Reported by: mavince 2007-10-05 16:44 +0000 [r84783] Russell Bryant * channels/chan_zap.c: Do deadlock avoidance in a couple more places. You can't lock two channels at the same time without doing extra work to make sure it succeeds. (closes issue #10895, patch by me) 2007-10-05 Russell Bryant * Asterisk 1.4.12.1 released. (This is mainly to include the app_queue fix for a memory leak on reload, but includes a couple of other bug fixes, as well.) 2007-10-05 01:39 +0000 [r84742] Russell Bryant * main/manager.c: Fix a copy/paste error in the description of UpdateConfig that was pointed out by JerJer on #asterisk-dev 2007-10-04 21:57 +0000 [r84692] Mark Michelson * apps/app_queue.c: Don't allocate space for queue members unless it's needed. You end up deleting dynamic members on a reload. Not good. closes issue (#10879, reported by dazza76, patched by me) 2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming * channels/chan_zap.c: callers of sig2str already add the word 'signalling' in the appropriate place, so don't duplicate it 2007-10-04 14:51 +0000 [r84637] Joshua Colp * apps/app_queue.c: Create a duplicate of the channel's member name as the tab completion stuff will free it. (closes issue #10884) Reported by: adamg 2007-10-03 22:59 +0000 [r84581] Tilghman Lesher * main/rtp.c: When an RFC 2833 event is sent that we don't recognize, ignore it, don't queue a NULL digit (closes issue #10877) 2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call in a switch expr doesn't execute correctly, because the commas in the function args are not converted to vertbars before the func is called. I modified just the switch code to convert the commas to vertbars if there, but if more of these sort of probs are found, I may have to resort to something a little more fundamental. We'll see, I guess. * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a null input to a switch statement results in a hangup; since switch is implemented with extensions, and the default case is implemented with a '.', and the '.' matches 1 or more remaining characters, the case where 0 characters exist isn't matched, and the extension isn't matched, and the goto fails, and a hangup occurs. Now, when a default case is generated, it also generates a single fixed extension that will match a null input. That extension just does a goto to the default extension for that switch. I played with an alternate solution, where I just tack an extra char onto all the patterns and the goto, but not the default case's pattern. Then even a null input will still have at least one char in it. But it made me nervous, having that extra char in , even if that's a pretty secret and low-level issue. 2007-10-02 Russell Bryant * Asterisk 1.4.12 released. 2007-10-02 20:06 +0000 [r84474] Russell Bryant * Makefile, build_tools/prep_tarball: * Don't build the menuselect-tree for the tarball, as it requires running the configure script first * Change the Makefile to note that menuselect-tree depends on the configure script. 2007-10-02 19:01 +0000 [r84410-84437] Jason Parker * res/res_features.c: Fix some odd formatting I missed.. * res/res_features.c: Finish up on transferee channel before return on failure. Issue 10821, patch by Ivan 2007-10-02 14:12 +0000 [r84370] Russell Bryant * channels/chan_sip.c: Use snprintf instead of sprintf in one place. There is no vulnerability here due to various buffer sizes around the code, but I still didn't like seeing a non length-limited copy of data coming off of the wire into a stack buffer, as this would be a problem in the future if buffer sizes elsewhere got changed or size limitations removed ... 2007-10-02 09:48 +0000 [r84345] Christian Richter * channels/chan_misdn.c: terminate USERUSER String with 0 2007-10-01 21:52 +0000 [r84291] Jason Parker * Makefile, Makefile.rules, channels/Makefile: Add dist-clean support for subdirs. Change h323 to only remove the Makefile on a dist-clean, rather than a clean. This fixes a bug I found with trying to run make after a make clean 2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard * main/channel.c, main/manager.c, channels/chan_agent.c: moved get_base_channel() code from action_redirect to ast_channel_masquerade() for issue 7706 and BE-160 2007-10-01 21:18 +0000 [r84273] Steve Murphy * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy... 2007-10-01 21:07 +0000 [r84271] Russell Bryant * main/utils.c, include/asterisk/lock.h: Fulfull a feature request from Qwell on the "core show locks" output. It will now note the lock type for each lock that a thread holds. (mutex, rdlock, or wrlock) 2007-10-01 20:27 +0000 [r84239] Steve Murphy * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue #10777 -- by returning a null for the parse tree when there's really nothing there, and making sure we don't try to do checking on a null tree. 2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant * res/res_agi.c: Add another sanity check in the AGI read loop. We really don't care about EAGAIN unless we didn't read an entire line. If there is a newline at the end if the read buffer, break, because we got the whole thing. (reported and patched by bmd) * include/asterisk/lock.h: Show rwlocks in the "core show locks" output. Before, it only showed mutexes. * channels/Makefile: Remove another file in "make clean". (closes issue #10814, paravoid) * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit. 2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp * configs/usbradio.conf.sample (removed): Remove chan_usbradio config file from tree, it is not present in here. (closes issue #10839) Reported by: casper * res/res_musiconhold.c: Fix randomness. save_pos was being set to 0 initially instead of -1, causing it to jump to position 0 when moh started. (closes issue #10859) Reported by: jamesgolovich Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich (license 176) * apps/app_dial.c: Only attempt early bridging if the options given to Dial() permit it. (closes issue #10861) Reported by: peekyb 2007-09-30 20:02 +0000 [r84146] Russell Bryant * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++ modules. The load and reload parameters were in the wrong place. (closes issue #10846, alebm) 2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added), pbx/ael/ael-test/ael-ntest22/t1/b.ael (added), pbx/ael/ael-test/ael-ntest22/t1/c.ael (added), pbx/ael/ael-test/ael-ntest22/t2/d.ael (added), pbx/ael/ael-test/ael-ntest22/t2/e.ael (added), pbx/ael/ael-test/ael-ntest22/t2/f.ael (added), pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22 (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added), pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ael-ntest22/t3/h.ael (added), pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ael-ntest22/t3/i.ael (added), pbx/ael/ael-test/ael-ntest22/t3/j.ael (added), pbx/ael/ael-test/ael-ntest22/qq.ael (added), pbx/ael/ael-test/ael-ntest22/t1 (added), pbx/ael/ael-test/ael-ntest22/t2 (added), pbx/ael/ael-test/ael-ntest22/t3 (added), pbx/ael/ael-test/ael-ntest22/extensions.ael (added), pbx/ael/ael-test/ael-ntest22 (added): This is a regression update that matches what I did in 84134 for AEL regressions. * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes 10786; All config files support #include with globbing (you know, *,[chars],?,{list,list},etc), so I've updated the AEL system to support this also. 2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher * main/say.c: Correct pronunciations of numbers for .nl (Closes issue #10837) * main/channel.c: Avoid a deadlock with ALL of the locks in the masquerade function, not just the pairs of channels. (Closes issue #10406) 2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard * main/manager.c, channels/chan_agent.c, include/asterisk/channel.h: if an Agent is redirected, the base channel should actually be redirected. This was causing multiple issues, especially issue 7706 and BE-160 2007-09-27 00:01 +0000 [r83976] Russell Bryant * pbx/pbx_dundi.c: remove a todo item that has been completed 2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming * channels/chan_alsa.c: avoid the weird usage of assert() in the ALSA header files that gcc 4.2 wants to complain about 2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant * channels/chan_sip.c: I changed my mind ... I think this should be a LOG_NOTICE. * channels/chan_sip.c: Add a log message that was requested by the masses in the developer tutorial session at Astricon. chan_sip did not output any message when a call was rejected because the extension was not found. This adds a verbose message (at verbose level 3) to note when this happens. * channels/chan_misdn.c: Fix building chan_misdn under dev-mode. (please run the configure script with --enable-dev-mode so this doesn't happen again ...) 2007-09-26 18:35 +0000 [r83879] Tilghman Lesher * channels/chan_zap.c: Remove unused 4k of memory on the program stack (closes issue #10827) 2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher * main/app.c: jmls pointed out that unsetting the group and setting the group to the blank string aren't quite the same. * build_tools/make_defaults_h: In the source, keys are relative to the datadir, not varlib (which is the same in most cases, but it's good to be accurate). Closes issue #10811 * doc/realtime.txt: Oops. Removed the unworkable workaround. This note should never have been in the release. * main/app.c: Making change to group splitting, as discussed on the -dev list. The main effect of this will be to permit Set(GROUP([cat])=), i.e. unsetting a group. 2007-09-24 07:54 +0000 [r83620] Christian Richter * channels/chan_misdn.c: fixed round_robin group dial method, this never worked well on BRI Ports (2 channels) 2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes are made here for the first arg in the for(arg1; arg2; arg3) {} statement, as were done for the 3rd arg. It can now be an assignment that will embedded in a Set() app, or a macro call, or an app call. * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for statement is now wrapped in Set() only if there's an '=' in that string. Otherwise, if it begins with '&', then a Macro call is generated; otherwise it is made into an app call. A bit more accomodating, keeps the new guys happy, and the guys with ael-1 code should be happy, too 2007-09-21 14:37 +0000 [r83432] Russell Bryant * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c, channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c, main/file.c, include/asterisk/sched.h, channels/chan_h323.c, pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl, channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y, main/db1-ast/hash/hash.c, include/asterisk/channel.h, channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, patch from qwell) 2007-09-21 13:34 +0000 [r83400] Joshua Colp * channels/chan_sip.c: Fix video under certain circumstances. It would have been possible for the formats on the channel to not contain the video format. (closes issue #10782) Reported by: cwhuang 2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant * main/asterisk.c: When daemonizing, don't change working directory to "/". It makes it not be able to do a core dump when not running as uid=root. (closes issue #10766, xrg) * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly ask for /bin/bash, as it uses bashisms. (closes issue #10772, reported by culrich) 2007-09-20 17:09 +0000 [r83246] Jason Parker * apps/app_disa.c: If # is pressed after dialing an extension in DISA, stop trying to collect more digits. (issue #10754) Reported by: atis Patches: app_disa.c.branch.patch uploaded by atis (license 242) app_disa.c.trunk.patch uploaded by atis (license 242) 2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp * channels/chan_sip.c: Make sure the minimum T1 timer value is obeyed in all cases. (closes issue #10768) Reported by: flefoll Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll (license 244) * channels/chan_sip.c: Fix a minor spelling error. (closes issue #10769) Reported by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll (license 244) 2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant * apps/app_system.c: The System() and TrySystem() applications can take a substantial amount of time to execute while not servicing the channel. So, put the channel in autoservice while the command is being executed. (closes issue #10726, reported by mnicholson) * funcs/func_curl.c: Using curl can take a substantial amount of time, so the channel should be autoserviced while waiting for it to complete. (closes issue #10725, reported by mnicholson) * channels/chan_iax2.c: When handling a reload of chan_iax2, don't use an ao2_callback() to POKE all peers. Instead, use an iterator. By using an iterator, the peers container is not locked while the POKE is being done. It can cause a deadlock if the peers container is locked because poking a peer will try to lock pvt structs, while there is a lot of other code that will hold a pvt lock when trying to go lock the peers container. (reported to me directly by Loic Didelot. Thank you for the debug info!) * main/manager.c: Fix up another potential race condition. Do the loop decrementing use count on events with the eventq protected from being changed. (reported on IRC by Ivan) 2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp * apps/app_queue.c: Protect the CDR record from modification by pbx_exec so that the application data contains the Queue data. (closes issue #10761) Reported by: snar Patches: app-queue-mixmonitor.patch uploaded by snar (license 245) * channels/chan_sip.c: (closes issue #10760) Reported by: dimas Patches: chan_sip.patch uploaded by dimas (license 88) Read in subscribecontext option in general to be the default. 2007-09-19 09:32 +0000 [r83023-83024] Christian Richter * channels/chan_misdn.c: removed comment which violates the coding guidelines. * channels/misdn_config.c, channels/chan_misdn.c, channels/misdn/chan_misdn_config.h: added 'astdtmf' option to allow configuring the asterisk dtmf detector instead of the mISDN_dsp ones. also added the patch from irroot #10190, so that dtmf tones detected by the asterisk detector are passed outofband to asterisk, to make any use of dtmf tones at all. 2007-09-19 00:19 +0000 [r82992] Russell Bryant * apps/app_flash.c: Change the description of app_flash to note how it can be a useful tool instead of just saying that it is generally a worthless feature. (Thanks to Jim Van Meggelen for pointing it out and providing the proposed text) 2007-09-18 23:41 +0000 [r82961] Joshua Colp * apps/app_queue.c: Initialize a variable to NULL to make the world happy. 2007-09-18 22:42 +0000 [r82929] Russell Bryant * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle interrupting the fgets() call when using FastAGI. This version of the patch maintains the original behavior of the code when not using FastAGI. (closes issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight mods by me Tested by: juggie, festr 2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey * main/manager.c: Corrected patch applied in revision r82887. * main/manager.c: Fixed a bug where http manager sessions prevented the eventq from being cleaned out because http manager sessions do not have a valid file descriptor. 2007-09-18 20:56 +0000 [r82867] Russell Bryant * main/manager.c: Fix a memory leak that can occur on systems under higher load. The issue is that when events are appended to the master event queue, they use the number of active sessions as a use count so it will know when all active sessions at the time the event happened have consumed it. However, the handling of the number of sessions was not properly synchronized, so the use count was not always correct, causing an event to disappear early, or get stuck in the event queue for forever. (closes issue #9238, reported by bweschke, patch from Ivan, modified by me) 2007-09-18 20:09 +0000 [r82865] Mark Michelson * apps/app_queue.c: Moving the logic for handling an empty membername to the create_member function so that there is a common place where this occurs instead of being spread out to several different places. 2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming * apps/app_queue.c: there is no need for conditional logic to select ->interface or ->membername, snince ->membername will always be populated 2007-09-18 16:31 +0000 [r82802] Russell Bryant * pbx/pbx_dundi.c: When copying the contents from the wildcard peer, do a deep copy instead of shallow copy so that it doesn't crash when beging destroyed. (closes issue #10546, patch by me) 2007-09-18 15:28 +0000 [r82751] Jason Parker * configs/sip.conf.sample: Correct the allowexternaldomains option in SIP sample config. Issue 10753 2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in ast_localtime() instead of a couple places in app_voicemail to prevent the problem everywhere instead of just a couple of places. (related to issue #10746) * apps/app_voicemail.c: Initialize some memory to fix crashes when leaving voicemail. This problem was fixed by running Asterisk under valgrind. (closes issue #10746, reported by arcivanov, patched by me) *** IMPORTANT NOTE: We need to check to see if this same bug exists elsewhere. * res/res_features.c: Handle the case where there are multiple dynamic features with the same digit mapping, but won't always match the activated on/by access controls. In that case, the code needs to keep trying features for a match. (reported by Atis on the asterisk-dev list, patched by me) 2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming * channels/chan_iax2.c: revert a change that wasn't supposed to be committed... doh! * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places where a logical member name (if specified) was not used, but instead the direct interface was listed 2007-09-17 02:00 +0000 [r82514] Joshua Colp * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead of passing a NULL pointer into snprintf pass "". It makes Solaris much happier. 2007-09-14 21:19 +0000 [r82444] Steve Murphy * main/cdr.c: closes issue #10668; thanks to arkadia for his patch; had to leave out the bit about ending the previous cdr in the fork; it would destroy current implementations. 2007-09-14 21:17 +0000 [r82435] Russell Bryant * configs/zapata.conf.sample: Add a note to help clarify the value set with the echocancel option. (inspired by Malcolm's blog post on blogs.digium.com about HPEC) 2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson * apps/app_queue.c: Crap, I broke the build. Fixed. * apps/app_queue.c: Adding member name field to manager events where they were missing before (closes issue #10721, reported by snar) 2007-09-14 17:48 +0000 [r82394] Jason Parker * channels/chan_zap.c: If a channel does not have an owner, do not try to set a channel variable. This will end up making the channel variable global, which is not right. Closes issue #10720, patch by flefoll. 2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add checking for libusb here, so nobody has to deal with conflicts in the chan_usbradio-1.4 branch every time the configure script gets changed * channels/chan_usbradio.c (removed), channels/xpmr (removed), channels/Makefile: Remove chan_usbradio from the main 1.4 branch. It can't live here because we have a strict policy to not include new features in release branches. However, I'm going to merge it into trunk, and I also have a special 1.4 based branch that includes this module. svn co http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4 2007-09-14 14:42 +0000 [r82376] Mark Michelson * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines (closes issue #10717, reported and patched by leedm777) 2007-09-14 01:24 +0000 [r82368] Jim Dixon * apps/app_rpt.c: Fixed problem with changes made to cdr functionality 2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming * channels/chan_usbradio.c: this new driver may not live in this branch for long (since it is a new feature), but it definitely should not be built by default 2007-09-14 00:34 +0000 [r82366] Jim Dixon * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added), channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added), channels/xpmr (added), channels/xpmr/LICENSE (added), channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added channel driver for USB Radio device and support thereof. 2007-09-13 23:11 +0000 [r82358] Jason Parker * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime 2007-09-13 20:16 +0000 [r82346] Mark Michelson * apps/app_queue.c: Preemptively fixing a possible segfault. It is possible that queuename is NULL (meaning pause ALL queues), so use q->name instead. 2007-09-13 20:11 +0000 [r82344] Jason Parker * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when mutliple threads tried to close the same file. Do we actually need the locking here? What happens if you open the same file twice, and two threads try to write to it at the same time? Is fputs() going to write out the entire line at once? I suspect that it could be possible for the second fopen to run during the first fputs, so the position could be in the middle of the previously written line... Issue 10347, initial patch by explidous (but I removed all of the paranoia stuff..) 2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant * main/astobj2.c: resolve a warning when not building under dev mode * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile in tracking astobj2 statistics if dev-mode is enabled. Also, when dev mode is enabled, register the CLI command that can be used to run the astobj2 test and print out statistics. 2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming * /, LICENSE: Merged revisions 82334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license exceptions ........ 2007-09-13 16:25 +0000 [r82326] Mark Michelson * apps/app_queue.c: Added logic to handle the unlikely case that someone has two queues with the same name. Asterisk will log a warning message letting the user know that one was already defined with that name and is it skipping all further instances. This also will work for realtime queues but in order for that to happen, the user would have to trigger a perfectly timed reload as a realtime queue is being looked up, which is highly unlikely (but taken care of nonetheless). 2007-09-13 11:47 +0000 [r82309] Philippe Sultan * channels/chan_gtalk.c: Closes issue #9401, reported and patched by irrot, with slight modifications by me. Handle DTMF sent by Asterisk properly. 2007-09-12 21:56 +0000 [r82296] Russell Bryant * res/res_agi.c: Fix a check of the wrong pointer, as pointed out by an XXX comment left in the code. The problem was harmless, however. 2007-09-12 21:28 +0000 [r82291] Tilghman Lesher * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone files 2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard * apps/app_meetme.c: remove a race condition for the creation of recordthread's, and fix a small memory leak. This closes issue# 10636 2007-09-12 20:12 +0000 [r82285] Tilghman Lesher * main/stdtime/private.h, main/stdtime/tzfile.h, include/asterisk/localtime.h, main/stdtime/localtime.c: Working on issue #10531 exposed a rather nasty 64-bit issue on ast_mktime, so we updated the localtime.c file from source. Next we'll have to write ast_strptime to match. 2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant * main/asterisk.c: Clean up the output of "asterisk -h". This tweaks the wording and wraps lines at 80 characters. (closes issue #10699, seanbright) * res/res_agi.c: revert patch from issue #10553, as someone not using fastagi reported that this broke their system. 2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson * apps/app_voicemail.c: Accidentally committed changes to app_voicemail which do NOT need to be in the 1.4 branch yet. reverting... * apps/app_voicemail.c, apps/app_queue.c: We should only initialize a realtime queue when it is allocated, not every time we access it. This prevents the members ao2_container from being reallocated every time the queue is accessed. I also removed a debug message I had accidentally left in on a previous commit. 2007-09-11 22:37 +0000 [r82267] Russell Bryant * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of these calls was reading bogus memory ... 2007-09-11 21:41 +0000 [r82265] Joshua Colp * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue #10679) Reported by: andrew Build under dev mode when K6OPTS is enabled. 2007-09-11 20:49 +0000 [r82263] Russell Bryant * apps/app_queue.c: Fix another missing unref of member objects. This one was pointed out by Marta. When building the outgoing list in try_calling(), a member reference is stored in each outgoing entry. However, when this list got destroyed, the reference was not released. 2007-09-11 20:36 +0000 [r82261] Steve Murphy * main/cdr.c: this change should fix issue # 10659 -- what I worry about is how many other bug reports it may generate. Hopefully, we can please the/a majority. Hopefully. We shall see. Calls not marked ANSWERED and with only one channel name will not be posted. This should eliminate the double CDR's. 2007-09-11 16:05 +0000 [r82252] Mark Michelson * apps/app_queue.c: All instances of ao2_iterators which were just named 'i' have been renamed to 'mem_iter' so that when refcounted queues are merged into trunk, there will be little confusion regarding iterator names, especially when a queue and member iterator are used in the same function. 2007-09-11 16:03 +0000 [r82250] Russell Bryant * pbx/pbx_dundi.c: The sample dundi.conf claims support for a wildcard peer entry - [*], but the code did not support it. This patch makes it work. (closes issue #10546, patch by dds, with some changes by me) 2007-09-11 16:01 +0000 [r82249] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a hold/retrieve issue. 2007-09-11 15:26 +0000 [r82245] Russell Bryant * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches: res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by: juggie When using fastagi, fgets() can return before a full line is read. Add explicit handling for the case where it gets interrupted. 2007-09-11 14:56 +0000 [r82243] Joshua Colp * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich (license 176) Don't leak memory when unloading DUNDi. 2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant * apps/app_queue.c: Add a couple more missing unrefs of queue member objects * apps/app_queue.c: Add a missing unref of a queue member in an error handling block * apps/app_queue.c: Document why membercount can not simply be replaced by ao2_container_count() * main/astobj2.c: backport astobj2 race condition fix. This function is the exact same as trunk so it applies here as well. 2007-09-10 18:02 +0000 [r82155] Tilghman Lesher * apps/app_queue.c: Convert struct member to use refcounts (closes issue #10199) 2007-09-10 15:02 +0000 [r82091] Mark Michelson * configs/misdn.conf.sample: Removing non-existent options from misdn configuration sample. (closes issue #10678, reported and patched by IgorG) 2007-09-09 02:35 +0000 [r82028] Tilghman Lesher * include/asterisk/lock.h: Fix inline compiles on really old compilers (who uses gcc 2.7 anymore, really?) 2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit() did not free the entry it removed. * .cleancount: (closes issue #10672) Bump the cleancount so that a "make clean" will be forced. This is needed because my fix in revision 81599 made a change to a data structure in file.h, and since file dependency tracking is only on with dev-mode enabled, file format modules that don't get rebuilt may crash, as is the case with this issue. This makes me wonder - how much faster does the code build without the file dependency tracking enabled? If it doesn't make much of a difference, then it may be worth just keeping it on all of the time, or perhaps just not in release tarballs, so that this type of issue is avoided. 2007-09-07 19:48 +0000 [r81923] Jason Parker * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used as the mixmonitor filename. This moves the setting of the MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch by sim. 2007-09-07 15:25 +0000 [r81886] Mark Michelson * configs/queues.conf.sample: Moving the explanation for joinempty to a more appropriate place 2007-09-06 22:28 +0000 [r81832] Russell Bryant * channels/chan_sip.c: (closes issue #9724, closes issue #10374) Reported by: kenw Patches: 9724.txt uploaded by russell (license 2) Tested by: kenw, russell Resolve a deadlock that occurs when doing a SIP transfer to parking. I come across this type of deadlock fairly often it seems. It is very important to mind the boundary between the channel driver and the core in respect to the channel lock and the channel-pvt lock. Channel drivers lock to lock the pvt and then the channel once it calls into the core, while the core will do it in the opposite order. The way this is avoided is by having channel drivers either release their pvt lock while calling into the core, or such as in this case, unlocking the pvt just long enough to acquire the channel lock. 2007-09-06 22:05 +0000 [r81778-81826] Jason Parker * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS for ASTLDFLAGS. This adds LDOPTS * include/asterisk/astobj2.h: This should fix a build issue that people building against uClibc were seeing with the addition of astobj2 2007-09-06 19:40 +0000 [r81776] Joshua Colp * apps/app_meetme.c: (closes issue #10122) Reported by: stevefeinstein Patches: meetme-unmute-manager.diff uploaded by qwell (license 4) Tested by: stevefeinstein After looking over the code I agree with Qwell. Setting the file descriptor to conference each time just causes a fight back and forth. 2007-09-06 16:56 +0000 [r81743] Philippe Sultan * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string length fixes. Removed an unused variable in aji_client structure (context) 2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson * apps/app_queue.c: Fixes an issue where valid DTMF had to be pressed twice to exit a queue if a member's phone was ringing. (closes issue #10655, reported by strider2k, patched by me) * res/res_features.c: Fixes a memory leak (closes issue #10658, reported and patched by Ivan) 2007-09-06 14:20 +0000 [r81650] Philippe Sultan * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 - and Google's XMPP server complaint, if set, the 'from' attribute must be set to the user's full JID. 2007-09-05 20:53 +0000 [r81599] Russell Bryant * include/asterisk/file.h, main/say.c, res/res_features.c, main/file.c, include/asterisk/channel.h: Fix an issue that can occur when you do an attended transfer to parking. If you complete the transfer before the announcement of the parking spot finishes, then the channel being parked will hear the remainder of the announcement. These changes make it so that will not happen anymore. Basically, res_features sets a flag on the channel is playing the announcement to so that the file streaming core knows that it needs to watch out for a channel masquerade, and if it occurs, to abort the announcement. (closes BE-182) 2007-09-05 17:18 +0000 [r81569] Tilghman Lesher * include/asterisk/lock.h: Solaris x86 compatibility fix 2007-09-05 15:19 +0000 [r81525] Mark Michelson * apps/app_queue.c: Fixing the build... 2007-09-05 15:14 +0000 [r81523] Jason Parker * channels/chan_phone.c: Do not try to unregister a NULL channel tech. Also changed load_module function to use defines rather than numbers for return values. Issue 10651, patch by rbraun_proformatique, with additions by me. 2007-09-05 15:03 +0000 [r81520] Mark Michelson * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to only count members who are logged in and available. (related to issue #10652, reported by wuwu) 2007-09-05 13:11 +0000 [r81492] Joshua Colp * main/channel.c: (closes issue #10650) Reported by: tacvbo Only print out that the spy was removed while holding the spy lock. 2007-09-04 20:54 +0000 [r81453-81455] Jason Parker * apps/app_followme.c: Rather than attempt to play a file, we can just check whether it exists. Issue 10634, patch by me, testing by pabelanger, sanity checked by bweschke * configs/followme.conf.sample: Change default followme config file to point to the correct files. Issue 10644, patch by pabelanger 2007-09-04 18:37 +0000 [r81448] Russell Bryant * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c: Remove the typedefs on ao2_container and ao2_iterator. This is simply because we don't typedef objects anywhere else in Asterisk, so we might as well make this follow the same convention. 2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming * channels/chan_sip.c: there is no point in sending 401 Unauthorized to a UAS that sent us a properly-formatted Authentication header with the expected username and nonce but an incorrect response (which indicates the shared secret does not match)... instead, let's send 403 Forbidden so that the UAS doesn't retry with the same authentication credentials repeatedly 2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp * channels/chan_iax2.c: (closes issue #10632) Reported by: jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich (license 176) Fix memory leak when unloading chan_iax2. The firmware files were not being freed. * main/channel.c: (closes issue #10476) Reported by: mdu113 Only look for the end of a digit when waiting for a digit. This in turn disables emulation in the core. * main/dns.c: (closes issue #10610) Reported by: john Patches: dns.c.patch uploaded by john (license 218) Tested by: mvanbaak Don't return a match if no SRV record actually exists. 2007-09-03 18:57 +0000 [r81433] Russell Bryant * channels/chan_iax2.c: Remove a couple of calls to ast_string_field_free_pools() on peers in error handling blocks in the code for building peers. The peer object destructor does this and doing it twice will cause a crash. (closes issue #10625, reported by and patched by pnlarsson) 2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson * apps/app_queue.c: Changed a comment to be more accurate. (really this is just a test to make sure I can commit properly from home) * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr into ao2_match_by_addr and making it available everywhere since it could be a handy callback to have 2007-08-31 21:27 +0000 [r81418] Russell Bryant * include/asterisk/astobj2.h: Remove references to a debugging parameter that does not exist 2007-08-31 19:48 +0000 [r81416] Mark Michelson * apps/app_queue.c: Fixed broken behavior of a reload on realtime queues. Prior to this patch, if a reload was issued and a realtime queue had callers waiting in it, then the queue would be removed from the queue list, but it would not actually be freed (in fact, a debug message warning about a memory leak would come up). With this patch, reloads do not touch realtime queues at all. 2007-08-31 19:16 +0000 [r81415] Tilghman Lesher * funcs/func_logic.c: The IF() function was not allowing true values that had embedded colons (closes issue #10613) 2007-08-31 18:44 +0000 [r81412] Jason Parker * apps/app_dial.c: Re-order dial options to be in line with the existing alpha order. Issue 10621, initial patch by junky 2007-08-31 17:38 +0000 [r81410] Philippe Sultan * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command available. Closes issue 10548, reported by keepitcool. 2007-08-31 15:53 +0000 [r81406] Joshua Colp * res/res_speech.c: Make it the engine's responsible to check for the presence of results. 2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming * codecs/codec_zap.c: add missing "transcoder show" (and deprecated "show transcoder") CLI commands that were in 1.2 but never added to 1.4 2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp * res/res_features.c: (closes issue #10618) Reported by: dimas Don't pass through the stopped sounds frame.... just drop it. * res/res_features.c: (closes issue #10009) Reported by: dimas Don't output a bridge failed warning message if it failed because one of the channels was part of the masquerade process. That is perfectly normal. 2007-08-30 22:05 +0000 [r81397] Mark Michelson * apps/app_queue.c: Removing an extraneous (and possibly misleading) log message. Firstly, if the announce file isn't found, the streaming functions will report it. Secondly, not all non-zero returns from play_file mean that the announce file wasn't found. Positive return values simply mean that a digit was pressed (most likely to skip through the announcement). (closes issue #10612, reported and patched by dimas) 2007-08-30 21:23 +0000 [r81395] Joshua Colp * channels/chan_sip.c: (closes issue #10514) Reported by: casper Patches: chan_sip.c.80129.diff uploaded by casper (license 55) Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it to ever be that value. 2007-08-30 21:11 +0000 [r81392] Steve Murphy * main/cdr.c: via issue 10599, where 'CDR already initialized' messages are being generated. Since all channels will have an init'd CDR attached at creation time, this message is now particularly useless. Removed. 2007-08-30 15:38 +0000 [r81383] Russell Bryant * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING define. (closes issue #10559, paravoid) 2007-08-30 15:35 +0000 [r81381] Mark Michelson * apps/app_queue.c: Changed some manager event messages to reflect whether a queue member is a realtime member or not 2007-08-30 15:33 +0000 [r81379] Russell Bryant * configs/modem.conf.sample (removed), configs/enum.conf.sample, configs/extensions.ael.sample: Fix a typo, update a reload command, and remove an unused configuration file. (closes issue #10606, casper) 2007-08-30 14:53 +0000 [r81375] Joshua Colp * main/pbx.c: (closes issue #10603) Reported by: jmls Patches: pbx.diff uploaded by jmls (license 141) Backport changes from 81372. Add REASON dialplan variable for when an originated call fails and the failed extension is executed. 2007-08-30 14:43 +0000 [r81373] Christian Richter * channels/chan_misdn.c: Fixed some warnings. 2007-08-30 14:23 +0000 [r81369] Joshua Colp * res/res_features.c: (issue #10599) Reported by: dimas Handle the -1 control subclass during feature dialing (it indicates to stop sounds). 2007-08-30 08:31 +0000 [r81367] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe issue where a misdn_read would lock the channel, but read would not return because it blocks. later chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER which could result in a deadlock situation. misdn_read will now not block forever anymore, and we don't queue the ANSWER frame at all when we already was called with misdn_answer -> answer would be called twice. Also we don't explicitly send a RELEASE_COMPLETE on receiption of a RELEASE anymore, because mISDN does that for us, this resulted in a problem on some switches, which would block our port after some calls for a short while. 2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson * apps/app_queue.c: This patch, in essence, will correctly pause a realtime queue member and reflect those changes in the realtime engine. (issue #10424, reported by irroot, patch by me) This patch creates a new function called update_realtime_member_field, which is a generic function which will allow any one field of a realtime queue member to be updated. This patch only uses this function to update the paused status of a queue member, but it lays the foundation for persisting the state of a realtime member the same way that static members' state is maintained when using the persistentmembers setting * apps/app_queue.c: Changed some tabs to spaces 2007-08-29 15:57 +0000 [r81342] Russell Bryant * main/Makefile: If chan_h323 is not being built, don't use g++ to do the final link of Asterisk. (in response to a question on the asterisk-dev list) 2007-08-29 15:52 +0000 [r81340] Mark Michelson * apps/app_queue.c: This fix creates a more accurate way of detecting whether realtime members were deleted. (closes issue 10541, reported by Alric, patched by me) The REALLY nice things about this patch is that queue members now have a "realtime" field which will be true if the member is a realtime member. This means we can check this value prior to certain processing if it should ONLY be done for realtime members. 2007-08-29 14:13 +0000 [r81331] Joshua Colp * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack. 2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant * channels/chan_iax2.c: Change the message about receiving a mini-frame before the first full voice frame to a DEBUG message. * pbx/pbx_dundi.c: revert unintentional changes in rev 81226 * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian tones. (closes issue #7953, hanabana) 2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using res_config_mysql (closes issue #10573, reported by chrisvaughan, patch suggested by chrisvaughan as well) * apps/app_queue.c: Resolve a potential deadlock. In this case, a single queue is locked, then the queue list. In changethread(), the queue list is locked, and then each individual queue is locked. Under the right circumstances, this could deadlock. As such, I have unlocked the individual queue before locking the queue list, and then locked the queue back after the queue list is unlocked. * channels/chan_agent.c: DTMF begin frames should be ignored so that when an agent acks a call with the '#' key, he doesn't cause a queue's announce file to be interrupted. Also went ahead and did the same for the '*' key and for ending a call. (closes issue #10528, reported by deskhack, patched by me) 2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most of the memory it allocates on unload and should be fixed (when I'm not in the middle of something else ...). * pbx/pbx_dundi.c: explicity define a variable as a boolean * res/res_musiconhold.c: (closes issue #10419) Reported by: mustardman Patches: asterisk-mohposition.diff.txt uploaded by jamesgolovich (license 176) This patch fixes a few problems with music on hold. * Fix issues with starting at the beginning of a file when it shouldn't. * Fix the inuse counter to be decremented even if the class had not been set to be deleted when not in use anymore * Don't arbitrarily limit the number of MOH files to 255 2007-08-27 15:01 +0000 [r81012] Joshua Colp * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang Patches: chan_sip-ChannelReload-20080825.patch uploaded by jesselang (license 202) Remove an extra \r\n to make the ChannelReload event conform with every other event. 2007-08-27 14:55 +0000 [r81010] Mark Michelson * apps/app_queue.c: Found a case where the queue's membercount is off. It does not take into account dynamic members on a reload. 2007-08-27 13:20 +0000 [r80974] Joshua Colp * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct jitter value output in the CLI to be as expected. 2007-08-26 18:11 +0000 [r80932] Russell Bryant * channels/chan_iax2.c: Remove an extra signal_condition() for the scheduler thread. (closes issue #10564, patch from casper) 2007-08-25 17:37 +0000 [r80895] Russell Bryant * channels/chan_iax2.c: Fix some issues with the handling of the scheduler in chan_iax2. Most of the places that scheduled items to be executed by the scheduler thread did not signal the scheduler thread to wake up so that it could recalculate the time until the next action. These changes will make the scheduler thread more responsive and ensure that actions get executed as close to when intended as possible instead of it being possible for very long delays. 2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard * apps/app_zapateller.c: An empty string is an empty callerid ... so zap it. This closes issue #10502, which was pointed out by dswartz. Thank you, and may the swartz be with you 2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are available when Asterisk first starts, then don't give up on poking peers. Allow the poke to get rescheduled so that it will work once the dnsmgr is able to resolve the host. (closes issue #10521, patch by jamesgolovich) * main/dsp.c: Improve the debouncing logic in the DTMF detector to fix some reliability issues. Previously, this code used a shift register of hits and non-hits. However, if the start of the digit isn't clean, it is possible for the leading edge detector to miss the digit. These changes replace the flawed shift register logic and also does the debouncing on the trailing edge as well. (closes issue #10535, many thanks to softins for the patch) 2007-08-24 19:52 +0000 [r80818] BJ Weschke * apps/app_queue.c: A minor correction to the available logic of autofill. If a queue member is paused, they're not really "available" so don't count them as such. Somewhat related to issue #10155 2007-08-24 18:52 +0000 [r80789] Steve Murphy * main/cdr.c: From a complaint by jmls, I realize that the message in cdr_disposition is unnecessary. To get failure disposition, just return -1; no use having more than one case do that. 2007-08-24 15:51 +0000 [r80750] Mark Michelson * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail. 2007-08-24 15:41 +0000 [r80747] Tilghman Lesher * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with the code, instead of only in documentation (closes issue #10549) 2007-08-24 15:28 +0000 [r80722] Russell Bryant * utils/ael_main.c: Tweak the formatting of this MODULEINFO block. I think this would have caused a "*" to get in the menuselect-tree file. 2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy * utils/ael_main.c: This change addresses JerJer's complaint that aelparse builds and installs even if pbx_ael is unchecked in the menuselect stuff. * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6: backport of 80649, a fix to an unreported problem in the ael parser, that results in a crash on a 64bit machine 2007-08-24 11:42 +0000 [r80661] Philippe Sultan * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are answered too early, which results in CDRs wrongly stating that a call was ANSWERED when the calling party cancelled a call before before being established. We must not answer the call upon reception of a 'transport-accept' iq packet, but this packet still needs to be acknowledged, otherwise the remote peer would close the call (like in #8970). 2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without warnings * channels/chan_misdn.c: make chan_misdn compile without warnings 2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant * include/asterisk/features.h, res/res_features.c: When executing a dynamic feature, don't look it up a second time by digit pattern after we already looked it up by name. This causes broken behavior if there is more than one feature defined with the same digit pattern. (closes issue #10539, reported by bungalow, patch by me) * funcs/func_timeout.c: Revert very broken fix for issue #10540 ... none of these values take ms so I don't know what I was thinking * funcs/func_timeout.c: Fix func_timeout to take values in floating point so 1.5 actually means 1.5 seconds instead of being rounded. (closes issue #10540, reported by spendergrass, patch by me) 2007-08-23 17:14 +0000 [r80505-80507] Jason Parker * /: *sigh* * /: use autotagged externals 2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming * channels/chan_zap.c: report the actual channel number that was unregistered, instead of assuming that the interface list consists of channels 1 through with no gaps in the sequence 2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant * channels/chan_iax2.c: Fix some code where it was possible for a reference to a peer to not get released when it should. Thank you to Marta Carbone for pointing this out! * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c: This is a hack to maintain old behavior of chan_iax2. This ensures that if the peers and users are being stored in a linked list, that they go in the list in the same order that the older code used. This is necessary to maintain the behavior of which peers and users get matched when traversing the container. * res/res_agi.c: Revert res_agi fix that didn't quite work until we get it right ... * include/asterisk/astobj2.h: Add some more documentation on iterating ao2 containers. The documentation implies that is possible to miss an object or see an object twice while iterating. After looking through the code and talking with mmichelson, I have documented the exact conditions under which this can happen (which are rare and harmless in most cases). * main/astobj2.c: When converting this code to use the list macros, I changed it so objects are added to the head of a bucket instead of the tail. However, while looking over code with mmichelson, we noticed that the algorithm used in ao2_iterator_next requires that items are added to the tail. This wouldn't have caused any huge problem, but it wasn't correct. It meant that if an object was added to a container while you were iterating it, and it was added to the same bucket that the current element is in, then the new object would be returned by ao2_iterator_next, and any other objects in the bucket would be bypassed in the traversal. * channels/chan_sip.c: Don't crash when using realtime in chan_sip without an insecure setting in the database. (closes issue #10348, reported by link55, fixed by me) * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h, channels/chan_iax2.c: Merge changes from team/russell/iax_refcount. This set of changes fixes problems with the handling of iax2_user and iax2_peer objects. It was very possible for a thread to still hold a reference to one of these objects while a reload operation tries to delete them. The fix here is to ensure that all references to these objects are tracked so that they can't go away while still in use. To accomplish this, I used the astobj2 reference counted object model. This code has been in one of Luigi Rizzo's branches for a long time and was primarily developed by one of his students, Marta Carbone. I wanted to go ahead and bring this in to 1.4 because there are other problems similar to the ones fixed by these changes, so we might as well go ahead and use the new astobj if we're going to go through all of the work necessary to fix the problems. As a nice side benefit of these changes, peer and user handling got more efficient. Using astobj2 lets us not hold the container lock for peers or users nearly as long while iterating. Also, by changing a define at the top of chan_iax2.c, the objects will be distributed in a hash table, drastically increasing lookup speed in these containers, which will have a very big impact on systems that have a large number of users or peers. The use of the hash table will be made the default in trunk. It is not the default in 1.4 because it changes the behavior slightly. Previously, since peers and users were stored in memory in the same order they were specified in the configuration file, you could influence peer and user matching order based on the order they are specified in the configuration. The hash table does not guarantee any order in the container, so this behavior will be going away. It just means that you have to be a little more careful ensuring that peers and users are matched explicitly and not forcing chan_iax2 to have to guess which user is the right one based on secret, host, and access list settings, instead of simply using the username. If you have any questions, feel free to ask on the asterisk-dev list. * res/res_agi.c: Juggie in #asterisk-dev was reporting problems where fgets would return without reading the whole line when using fastagi. When this happens, errno was set to EINTR or EAGAIN. This patch accounts for the possibility and lets fgets continue in that case. 2007-08-22 18:53 +0000 [r80302-80330] Jason Parker * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h, build_tools/strip_nonapi, build_tools/prep_moduledeps, build_tools/make_buildopts_h: Fix a few build issues in Solaris (and likely others). Use GREP and ID variables from autoconf. Reported to me in #asterisk-dev I forgot who reported this - sorry. :( * Makefile: Change a syntax that the GNU make in Solaris dislikes. * build_tools/make_version: Fix a bashism (we explicitly request /bin/sh). Remove some oddly placed quotes I found in passing. 2007-08-22 16:21 +0000 [r80257] Russell Bryant * Makefile: Honor the contents of the COPTS variable as custom target CFLAGS. Apparently this is what openwrt does. (reported by Brian Capouch on the asterisk-dev list, patch by me) 2007-08-22 16:14 +0000 [r80255] Joshua Colp * main/rtp.c: (closes issue #10526) Reported by: sinistermidget Revert commit from issue #10355 and return timestamp skew to 640. 2007-08-21 Russell Bryant * Asterisk 1.4.11 released. 2007-08-21 18:42 +0000 [r80183] Russell Bryant * channels/chan_sip.c: Don't record SIP dialog history if it's not turned on. Also, put an upper limit on how many history entires will be stored for each SIP dialog. It is currently set to 50, but can be increased if deemed necessary. (closes issue #10421, closes issue #10418, patches suggested by jmoldenhauer, patches updated by me) (Security implications documented in AST-2007-020) 2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing the diffs from ulaw.h and alaw.h for now; accidentally added them in 80166 * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h: This patch solves problem 1 in 8126; it should not slow down the alaw codec, but should prevent signal degradation via multiple trips thru the codec. Fossil estimates the twice thru this codec will prevent fax from working. 4-6 times thru would result hearable, noticeable, voice degradation. 2007-08-21 15:22 +0000 [r80132] Russell Bryant * channels/chan_mgcp.c: Don't try to dereference the owner channel when it may not exist (issue #10507, maxper) 2007-08-21 15:03 +0000 [r80130] Jason Parker * configs/cdr.conf.sample: (issue #10510) Reported by: casper Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few errors in sample cdr config file. 2007-08-20 21:57 +0000 [r80088] Russell Bryant * apps/app_queue.c: Fix the build of app_queue 2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson * apps/app_queue.c: After a discussion on #asterisk-dev, it was decided that this should be in 1.4 as well. (issue #10424, reported and patched by irroot) * apps/app_queue.c: Found a pointless ternary if. member->dynamic was set to 1 and has no opportunity to change between then and this line, so "dynamic" will ALWAYS be output. 2007-08-20 16:08 +0000 [r80047] Jason Parker * configs/extensions.conf.sample: (issue #10499) Reported by: casper Patches: extensions.conf.sample.diff uploaded by casper (license 55) Update CLI examples in extensions.conf.sample to reflect command changes. 2007-08-20 15:34 +0000 [r80044] Mark Michelson * apps/app_voicemail.c: Ukrainian language voicemail support. (closes issue #10458, reported and patched by Oleh) 2007-08-20 02:42 +0000 [r79998] Tilghman Lesher * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by snuffy via IRC) 2007-08-18 14:30 +0000 [r79947] Tilghman Lesher * apps/app_voicemail.c: Don't allocate vmu for messagecount when we could just use the stack instead (closes issue #10490) Also, remove a useless (and leaky) SQLAllocHandle (closes issue #10480) 2007-08-17 21:01 +0000 [r79912] Russell Bryant * channels/chan_zap.c: Avoid a crash in the handling of DTMF based Caller ID. It is valid for ast_read to return NULL in the case that the channel has been hung up. (crash reported by anonymouz666 on IRC in #asterisk-dev) 2007-08-17 19:14 +0000 [r79906] Mark Michelson * apps/app_voicemail.c: Patch allows for more seamless transition from file storage voicemail to ODBC storage voicemail. If a retrieval of a greeting from the database fails, but the file is found on the file system, then we go ahead an insert the greeting into the database. The result of this is that people who switch from file storage to ODBC storage do not need to rerecord their voicemail greetings. 2007-08-17 19:12 +0000 [r79902-79904] Jason Parker * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h: Don't send a semicolon over the wire in sip notify messages. Caused by fix for issue 9938. I basically took the code that existed before 9938 was fixed, and copied it into a new function - ast_unescape_semicolon There should be very few places this will be needed (pbx_config does NOT need this (see issue 9938 for details)) Issue 10430, patch by me, with help/ideas from murf (thanks murf). * channels/chan_local.c: Re-add the setting of callerid name and number. Issue 10485, reported by and fix explained by paradise. 2007-08-17 13:37 +0000 [r79857] Russell Bryant * channels/chan_sip.c: Fix some crashes in chan_sip. This patch changes various places that add items to the scheduler to ensure that they don't overwrite the ID of a previously scheduled item. If there is one, it should be removed. (closes issue #10391, closes issue #10256, probably others, patch by me) 2007-08-17 08:22 +0000 [r79833] Christian Richter * channels/chan_misdn.c: sometimes we don't need to signal dtmf tones to asterisk, we just want them to go through as inband. Otherwise they might be generated by the other channel partner and then there is a double tone. 2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant * res/res_musiconhold.c: Fix a little race condition that could cause a crash if two channels had MOH stopped at the same time that were using a class that had been marked for deletion when its use count hits zero. * res/res_musiconhold.c: This patch fixes a bug where reloading the module with "module reload" did not delete classes from memory that were no longer in the config. This patch fixes that problem as well as another one. Previously, if you reloaded MOH using the "moh reload" CLI command, which behaved differently than "module reload ...", MOH had to be stopped on every channel and started again immediately. However, there was no way to tell what class was being used, so they would all fall back to the default class. (closes issue #10139) Reported by: blitzrage Patches: asterisk-10139-advanced.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were introduced by making frame handling and scheduling multi-threaded. Unfortunately, we have to do some expensive deadlock avoidance when queueing frames on to the ast_channel owner of the IAX2 pvt struct. This was already handled for regular frames, but ast_queue_hangup and ast_queue_control were still used directly. Making these changes introduced even more places where the IAX2 pvt struct can disappear in the context of a function holding its lock due to calling a function that has to unlock/lock it to avoid deadlocks. I went through and fixed all of these places to account for this possibility. (issue #10362, patch by me) 2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson * channels/chan_agent.c: Fixes a problem where agents would get stuck busy due to their wrapuptime being longer than the queue's wrapuptime and ringinuse=no for the queue. (closes issue #10215, reported by Doug, repaired by me) Special thanks to fkasumovic for pointing out the source of the problem and to bweschke for helping to come up with a solution! * apps/app_voicemail.c: base_encode is not trying to open a log file, so we should not call it a log file in the warning. (related to issue #10452, reported by bcnit) 2007-08-16 09:37 +0000 [r79665] Philippe Sultan * res/res_jabber.c: A fix for two critical problems detected while working with Daniel McKeehan in issue #10184. Upon priority change, the resource list is not NULL terminated when moving an item to the end of the list. This makes Asterisk endlessy loop whenever it needs to read the list. Jids with different resource and priority values, like in Gmail's and GoogleTalk's jabber clients put that problem in evidence. Upon reception of a 'from' attribute with an empty resource string, Asterisk crashes when trying to access the found->cap pointer if the resource list for the given buddy is not empty. This situation is perfectly valid and must be handled. The Gizmoproject's jabber client put that problem in evidence. Also added a few comments in the code as well as a handle for the capabilities from Gmail's jabber client, which are stored in a caps:c tag rather than the usual c tag. Closes issue #10184. 2007-08-16 08:21 +0000 [r79642] Christian Richter * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when we want to send IA5 Chars. 2007-08-15 14:40 +0000 [r79553] Joshua Colp * main/rtp.c: (closes issue #10440) Reported by: irroot (closes issue #10454) Reported by: flo_turc Increase maximum timestamp skew to 120. 20 was apparently far too low. 2007-08-15 14:26 +0000 [r79527] Mark Michelson * apps/app_voicemail.c: Fixed an error in the Russian language voicemail intro. (issue #10458, reported and patched by Oleh) 2007-08-15 14:18 +0000 [r79523] Joshua Colp * channels/chan_sip.c: (closes issue #10456) Reported by: irroot Patches: sip_timeout.patch uploaded by irroot (license 52) Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten. 2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant * channels/chan_iax2.c: Fix another spot where an iax2_peer would be leaked if realtime was in use. * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2 related to the use of realtime. I found these while working on iax2_peer object reference tracking. 2007-08-14 15:27 +0000 [r79397] Joshua Colp * res/res_features.c: (closes issue #10415) Reported by: atis Revert fix for #10327 as it causes more issues then it solves. 2007-08-13 22:40 +0000 [r79363] Steve Murphy * pbx/pbx_ael.c: memset really, really needs to be used here. 2007-08-13 21:57 +0000 [r79334] Joshua Colp * res/res_speech.c, apps/app_speech_utils.c, include/asterisk/speech.h: Instead of accepting a single DTMF character accept a full string. 2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant * channels/chan_iax2.c: Don't call find_peer in registry_authrequest with the pvt lock held to avoid a deadlock. * channels/chan_iax2.c: Release the pvt lock before calling find_peer in register_verify to avoid a deadlock. Also, remove some unnecessary locking in auth_fail that was only done recursively. * channels/chan_iax2.c: Don't call find_peer within update_registry with a pvt lock held. This can cause a deadlock as the code will eventually call find_callno. * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I have tracked them down to a single core issue. You can not call find_callno() while holding a pvt lock as this function has to lock another (every) other pvt lock. Doing so can lead to a classic deadlock. So, I am tracking down all of the code paths where this can happen and fixing them. The fix I committed earlier today was along the same theme. This patch fixes some code down the path of authenticate_reply. 2007-08-13 17:49 +0000 [r79255] Steve Murphy * pbx/ael/ael-test/ref.ael-vtest21 (added), pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ael-vtest21/extensions.ael (added), pbx/ael/ael-test/ael-vtest21 (added), pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch fixes bug 10411. I added a new regression test, some regression test cleanups 2007-08-13 15:28 +0000 [r79214] Russell Bryant * channels/chan_iax2.c: Fix a potential deadlock in socket_process. check_provisioning can eventually call find_callno. You can't hold a pvt lock while calling find_callno because it goes through and locks every single one looking for a match. 2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp * res/res_speech.c, apps/app_speech_utils.c, include/asterisk/speech.h: Add an API call to allow the engine to know that DTMF was received. * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c, channels/chan_local.c, channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_gtalk.c, channels/chan_iax2.c: (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. 2007-08-11 05:23 +0000 [r79142] Tilghman Lesher * res/res_odbc.c: Ensure the connection gets marked as used at allocation time (closes issue #10429, report and fix by mnicholson) 2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From a user complaint on #asterisk, I have forced pbx_spool to explain what reason codes mean, when they are logged * main/cdr.c: Re bug behavior mentioned in #asterisk, made this tweak to code, to prevent hundreds of log messages from being generated * main/cdr.c: This will help debug; from a question asked on #asterisk 2007-08-10 Russell Bryant * Asterisk 1.4.10.1 released. 2007-08-10 15:20 +0000 [r78995] Russell Bryant * include/asterisk/lock.h: The last set of changes that I made to "core show locks" made it not able to track mutexes unless they were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized with ast_mutex_init() were not tracked. It should work now. 2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp * main/file.c: Don't bother having the core pass through or emulate begin DTMF frames when in an ast_waitstream. It only cares about the end of DTMF. * configs/queues.conf.sample: (closes issue #10422) Reported by: bhowell Add note to sample configuration about module load order and how it can cause perfectly good queue members to be marked as invalid. 2007-08-10 13:24 +0000 [r78936] Christian Richter * channels/chan_misdn.c, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser information element. We send them now also in the disconnect message. 2007-08-09 23:47 +0000 [r78907] Mark Michelson * apps/app_voicemail.c: Improved a bit of logic regarding comma-separated mailboxes in has_voicemail. Also added some braces to some compound if statements since unbraced if statements scare me in general. 2007-08-09 23:10 +0000 [r78891] Steve Murphy * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty output 2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson * apps/app_voicemail.c: Removing some extra debug code I left in my last commit * apps/app_voicemail.c: Quite a few changes regarding IMAP storage. 1. instead of using inboxcount as the core message counting function, we use messagecount instead. This makes it possible to count messages in folders besides just INBOX and Old. 2. inboxcount and hasvoicemail now use messagecount as their means of determining return values. 3. Added a copy_message function for IMAP storage. Unfortunately I don't have the means to test it, but it seems like a pretty straightforward function. 4. Removed a #ifndef IMAP_STORAGE and matching #endif from leave_voicemail for a couple of reasons. One, we want to support copying mail to multiple IMAP boxes, and two, IMAP was broken because a STORE macro had been moved into this section of code. * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I put some new code in the wrong place and so I've reverted the canreinvite section to how it was and put my new code where it should be. 2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant * apps/app_voicemail.c: add a comment to indicate that inboxcount for ODBC_STORAGE needs to be fixed to support multiple mailboxes * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for ODBC_STORAGE. Also, leave a comment for this to be fixed for IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was working on this code right now for another reason. This is broken even worse in trunk, but for a different reason. The fact that the mailbox option supported multiple mailboxes is completely not obvious from the code in the channel drivers. Anyway, I will fix that in another commit ... * apps/app_meetme.c: Fix a problem with the combination of the 'F' option to pass DTMF through a conference and options that use DTMF to activate various features. The problem was that the BEGIN frame would be passed through, but the END frame would get intercepted to activate a feature. Then, the other conference members would hear DTMF for forever, which they didn't seem to like very much. (closes issue #10400, reported by stevefeinstein, fixed by me) 2007-08-08 19:29 +0000 [r78646] Jason Parker * doc/jabber.txt: Fix mogs email address. 2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson * apps/app_voicemail.c: Fixed some compiler warnings so that compiling with dev-mode and IMAP storage would not have any errors. This section of code may get changed again shortly since my change uncovers a rather silly bit of logic. * apps/app_queue.c: Changing a bit of logic so that someone will NEVER exit the queue on timeout unless they have enabled the 'n' option. This commit relates to issue #10320. Thanks to jfitzgibbon for detailing the idea behind this code change. 2007-08-08 13:51 +0000 [r78569] Joshua Colp * configs/sip.conf.sample: (closes issue #10335) Reported by: adamgundy Update sip.conf to include another scenario where directrtpsetup will fail. 2007-08-07 Russell Bryant * Asterisk 1.4.10 released. 2007-08-07 20:57 +0000 [r78488] Russell Bryant * res/res_config_odbc.c: Fix the build of this module on 64-bit platforms 2007-08-07 19:43 +0000 [r78450] Mark Michelson * apps/app_voicemail.c: The logic behind inboxcount's return value was reversed in has_voicemail and message_count. (closes issue #10401, reported by st1710, patched by me) 2007-08-07 19:34 +0000 [r78437] Tilghman Lesher * res/res_odbc.c: Don't free the environment handle when the connection fails, because other connections might be depending upon it 2007-08-07 19:11 +0000 [r78416] Jason Parker * channels/chan_sip.c: Allow chan_sip to build in devmode 2007-08-07 19:09 +0000 [r78415] Tilghman Lesher * apps/app_voicemail.c, res/res_config_odbc.c, apps/app_directory.c: Reconnection doesn't happen automatically when a DB goes down (fixes issue #9389) 2007-08-07 18:25 +0000 [r78375] Jason Parker * channels/chan_skinny.c: Properly check the capabilities count to avoid a segfault. (ASA-2007-019) 2007-08-07 17:45 +0000 [r78371] Russell Bryant * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) | 4 lines Revert patch committed for issue #9660. It broke E&M trunks. (closes issue #10360) (closes issue #10364) ........ 2007-08-06 21:41 +0000 [r78275] Joshua Colp * main/channel.c: Add additional DTMF log messages to help when debugging issues. 2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant * channels/chan_iax2.c: Fix an issue where dynamic threads can get free'd, but still exist in the dynamic thread list. (closes issue #10392, patch from Mihai, with credit to his colleague, Pete) * include/asterisk/linkedlists.h: Fix the return value of AST_LIST_REMOVE(). This shouldn't be causing any problems, though, because the only code that uses the return value only checks to see if it is NULL. (closes issue #10390, pointed out by mihai) 2007-08-06 16:32 +0000 [r78182] Joshua Colp * channels/chan_sip.c: It is possible for a transfer to occur before the remote device has our tag in which case they send none in the transfer. In this case we need to not fail the transfer dialog lookup. 2007-08-06 16:30 +0000 [r78180] Jason Parker * main/config.c: Fix an issue with using UpdateConfig (manager action) where escaped semicolons in a config would be converted to just semicolons (\; to ;) Issue 9938 2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed. * configure, configure.ac: (closes issue #10383) Reported by: rizzo Include stdlib.h so NULL gets defined for gethostbyname_r checks. 2007-08-06 13:33 +0000 [r78164] Mark Michelson * channels/chan_sip.c: Fixed a mistake I made in realtime_peer which caused it to return NULL every time. Thanks to Jon Fealy for emailing me the correction. 2007-08-05 14:18 +0000 [r78146] Tilghman Lesher * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes bug #10382) 2007-08-05 04:15 +0000 [r78143] Russell Bryant * include/asterisk/lock.h: Fix compilation failure when MALLOC_DEBUG is enabled, but DEBUG_THREADS is not 2007-08-05 03:29 +0000 [r78139] Tilghman Lesher * channels/chan_sip.c: If peer is not found, the error message is misleading (should be peer not found, not ACL failure) 2007-08-03 20:25 +0000 [r78103] Mark Michelson * main/config.c, channels/chan_sip.c, include/asterisk/config.h: Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers. Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the IP address. In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct 2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant * apps/app_voicemail.c: (closes issue #10194) Reported by: blitzrage Patches: bug0010194 uploaded by vovochka Tested by: blitzrage Fix a problem when you call Voicemail() with multiple mailboxes specified and ODBC_STORAGE is in use. The audio part of the message was only given to the first mailbox specified. * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some improvements to lock debugging. These changes take effect with DEBUG_THREADS enabled and provide the following: * This will keep track of which locks are held by which thread as well as which lock a thread is waiting for in a thread-local data structure. A reference to this structure is available on the stack in the dummy_start() function, which is the common entry point for all threads. This information can be easily retrieved using gdb if you switch to the dummy_start() stack frame of any thread and print the contents of the lock_info variable. * All of the thread-local structures for keeping track of this lock information are also stored in a list so that the information can be dumped to the CLI using the "core show locks" CLI command. This introduces a little bit of a performance hit as it requires additional underlying locking operations inside of every lock/unlock on an ast_mutex. However, the benefits of having this information available at the CLI is huge, especially considering this is only done in DEBUG_THREADS mode. It means that in most cases where we debug deadlocks, we no longer have to request access to the machine to analyze the contents of ast_mutex_t structures. We can now just ask them to get the output of "core show locks", which gives us all of the information we needed in most cases. I also had to make some additional changes to astmm.c to make this work when both MALLOC_DEBUG and DEBUG_THREADS are enabled. I disabled tracking of one of the locks in astmm.c because it gets used inside the replacement memory allocation routines, and the lock tracking code allocates memory. This caused infinite recursion. * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control frames when the mohinterpret option is set to "passthrough". This was pointed out by Kevin in the middle of a training session. * channels/chan_iax2.c: Don't reuse the timespec that was set to 0 in the previous timedwait as it will just return immediately. Also, fix some logic so the thread's lock isn't unlocked twice in the weird case of dynamic threads getting acquired right after a timeout. (pointed out by SteveK) 2007-08-02 21:53 +0000 [r77993-77996] Jason Parker * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we actually allow 6 chars to be sent. Also make note of the "A" option of date format. Issue 9779, modifications by DEA, wedhorn, and myself. * channels/chan_skinny.c: If a device disconnects, the session will go away. If this happens during call setup, we need to give up. Issue 10325. 2007-08-02 19:25 +0000 [r77949] Russell Bryant * channels/chan_iax2.c: Fix the case where a dynamic thread times out waiting for something to do during the first time it runs. This shouldn't ever happen, but we should account for it anyway. (pointed out by pete, who works with mihai) 2007-08-02 18:42 +0000 [r77947] Jason Parker * channels/chan_skinny.c: Make sure we clear the prompt status message on a hangup. Also rearrange messages to better fit with what a wireshark trace shows it should be. Issue 10299, initial patch and solution by sbisker, modified by me to fit with wireshark trace. 2007-08-02 18:21 +0000 [r77945] Steve Murphy * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 line This patch hopefully solves 10141; The user is running with it, and it doesn't appear to harm asterisk's operation, and may prevent a crash. I'll store it in 1.2, as we have shut down support on 1.2, but since I developed the patch before support finished, and it might affect 1.4 and trunk, I'm going ahead with it. ........ 2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant * channels/chan_iax2.c: Fix another race condition in the handling of dynamic threads. If the dynamic thread timed out waiting for something to do, but was acquired to perform an action immediately afterwords, then wait on the condition again to give the other thread a chance to finish setting up the data for what action this thread should perform. Otherwise, if it immediately continues, it will perform the wrong action. (reported on IRC by mihai, patch by me) (related to issue #10289) * channels/chan_iax2.c: Add another sanity check to vnak_retransmit(). This check ensures that frames that have already been marked for deletion don't get retransmitted. (closes issue #10361, patch from mihai) 2007-08-02 15:15 +0000 [r77890-77894] Jason Parker * channels/chan_skinny.c: Make sure that we show the correct extension if dialed from a macro "From: 5555" rather than "From: s" Issue 10358, initial patch by DEA, reworked by me to use S_OR, tested by sbisker * channels/chan_skinny.c: Put in some additional debug information for softkey/stimulus messages. Issue 10291, patch by DEA. 2007-08-01 22:16 +0000 [r77887] Russell Bryant * channels/chan_iax2.c: Fix some race conditions which have been causing weird problems in chan_iax2. The most notable problem is that people have been seeing storms of VNAK frames being sent due to really old frames mysteriously being in the retransmission queue and never getting removed. It was possible that a dynamic thread got created, but did not acquire its lock before the thread that created it signals it to perform an action. When this happens, the thread will sleep until it hits a timeout, and then get destroyed. So, the action never gets performed and in some cases, means a frame doesn't get transmitted and never gets freed since the scheduler never gets a chance to reschedule transmission. Another less severe race condition is in the handling of a timeout for a dynamic thread. It was possible for it to be acquired to perform at action at the same time that it hit a timeout. When this occurs, whatever action it was acquired for would never get performed. (patch contributed by Mihai and SteveK) (closes issue #10289) (closes issue #10248) (closes issue #10232) (possibly related to issue #10359) 2007-08-01 22:14 +0000 [r77886] Tilghman Lesher * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does not compile cleanly (missing def) 2007-08-01 21:08 +0000 [r77883] Jason Parker * channels/chan_skinny.c: Fix an issue that caused one-way audio on some newer devices (specifically the 7921), due to sending packets in the wrong order during hangup. Also make sure we clear tones/messages on the correct line/instance. Issue 10291, patch by DEA, tested by sbisker and myself. 2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp * main/cli.c: (closes issue #10351) Reported by: ftarz Some platforms don't like it when you pass NULL to vsnprintf so pass "" instead. * include/asterisk/threadstorage.h, channels/chan_mgcp.c, apps/app_voicemail.c, main/acl.c, utils/smsq.c, channels/chan_iax2.c: Add some fixes for building on Solaris. * main/utils.c: Whoops, I meant R_5 not R5. * configure, configure.ac: And for my last trick... make sure that if gethostbyname_r is exported by a library that it is used. * configure, include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Extend autoconf logic to determine which version of gethostbyname_r is on the system. 2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson * apps/app_queue.c: Fixes an issue I introduced to queues wherein a queue with joinempty=yes would kick people out of the queue because of erroneously thinking the 'n' option was in use. (closes issue #10320, reported by jfitzgibbon, patched by me, tested by blitzrage and me) Thank you blitzrage for all the testing you've done lately with queues! It's much appreciated! * apps/app_queue.c: If a queue uses dynamic realtime members, then the member list should be updated after each attempt to call the queue. This fixes an issue where if a caller calls into a queue where no one is logged in, they would wait forever even if a member logged in at some point. (closes issue #10346, reported by and tested by blitzrage, patched by me) 2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon * apps/app_rpt.c: Much newer version, 0.70 with much additions * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in logic in TONE_VERIFY/RELAX mode in chan_zap. 2007-07-31 20:59 +0000 [r77844] Steve Murphy * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1 line This probably isn't super-general, but it's a first stab at using kill -11 to generate a core file instead of gcore. ........ 2007-07-31 16:17 +0000 [r77831] Joshua Colp * res/res_speech.c, include/asterisk/speech.h: Add a flag to the speech API that allows an engine to set whether it received results or not. 2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without DEBUG_THREADS or it does nothing 2007-07-31 15:21 +0000 [r77824] Mark Michelson * channels/chan_sip.c: This patch makes Asterisk send 100 Trying provisional responses upon receipt of re-invites. This makes it so that if there are two or more Asterisk servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites. (closes issue #10274, reported by cstadlmann, patched by me with approval from file) 2007-07-30 20:17 +0000 [r77795] Jason Parker * main/say.c: Applications like SayAlpha() should not hang up the channel if you request an "unknown" character such as a comma. Instead, skip the character and move on. Issue 10083, initial patch by jsmith, modified by me. 2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant * channels/chan_iax2.c: Fix an issue that could potentially cause corruption of the global iax frame queue. In the network_thread() loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE macro. However, to remove an element of the list within this loop, it used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I believe could leave some of the internal variables of the SAFE macro invalid. Mihai says that he already made this change in his local copy and it didn't help his VNAK storm issues, but I still think it's wrong. :) * res/res_agi.c: (closes issue #10279) Reported by: seanbright Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71) Allow the "agi_network: yes" line to be printed out in the AGI debug output. Also, allow partial writes to be handled when writing out this line just like it is for all of the others. * main/channel.c: file and I both committed changes for issue #10301. Remove a duplicated assignment to restore the original value of the previous channel. 2007-07-30 18:43 +0000 [r77783] Tilghman Lesher * /, res/res_agi.c: Merged revisions 77782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) | 2 lines Revert change in revision 71656, even though it fixed a bug, because many people were depending upon the (broken) behavior. ........ 2007-07-30 17:29 +0000 [r77780] Russell Bryant * main/channel.c: (closes issue #10301) Reported by: fnordian Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) Additional changes by me Fix some problems in channel_find_locked() which can cause an infinite loop. The reference to the previous channel is set to NULL in some cases. These changes ensure that the reference to the previous channel gets restored before needing it again. I'm not convinced that the code that is setting it to NULL is really the right thing to do. However, I am making these changes to fix the obvious problem and just leaving an XXX comment that it needs a better explanation that what is there now. 2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp * res/res_features.c: (closes issue #10327) Reported by: kkiely Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place. * main/channel.c: (closes issue #10301) Reported by: fnordian Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function. * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 lines (closes issue #10334) Reported by: ramonpeek Pass through the return value from macro_exec through the MacroIf application. ........ 2007-07-27 18:15 +0000 [r77571] Tilghman Lesher * res/res_odbc.c: Missing newline 2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois Patches: cdr_pgsql.patch uploaded by prashant (license 114) Finish the Postgresql connection after the log messages are printed so we don't access invalid memory. * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99) Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing. 2007-07-27 14:30 +0000 [r77490] Mark Michelson * channels/chan_sip.c: "re-invite" was misspelled 2007-07-26 23:19 +0000 [r77460] Joshua Colp * main/channel.c: (closes issue #10302) Reported by: litnialex If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already. 2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming * doc/mp3.txt: change protocol for downloads as well * doc/mp3.txt, sounds/Makefile: use new canonical name for download server 2007-07-26 21:23 +0000 [r77410] Russell Bryant * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined in trunk, but not in 1.4. When Asterisk is compiled under dev mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to define it in the same way that trunk does. Also, revert the change that added this define in the Makefile The advantage to doing it this way is that buildopts.h gets installed when you install Asterisk. Then, when building any out of tree modules, or building asterisk-addons, these modules know which options the rest of Asterisk was built with. 2007-07-26 20:35 +0000 [r77380] Mark Michelson * Makefile, main/logger.c: Fixes to get ast_backtrace working properly. The AST_DEVMODE macro was never defined so the majority of ast_backtrace never attempted compilation. The makefile now defines AST_DEVMODE if configure was run with --enable-dev-mode. Also, changes were made to acccomodate 64 bit systems in ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for their roles in allowing me to get this committed 2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher * main/logger.c: Missed one * main/logger.c: Oops, that builtin define should be all-lowercase. 2007-07-26 18:30 +0000 [r77318] Mark Michelson * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur, meaning free gets called on the same variable twice. This patch sets the connection to NULL after calls to PQfinish so that the problem does not occur. Also in this patch, prashant_jois informed me that it is safe to pass a null pointer to PQfinish, so I have removed the check for conn's existence from my_unload_module. (closes issue 10295, reported by junky, patched by me with input from prashant_jois) 2007-07-25 22:39 +0000 [r77191] Steve Murphy * apps/app_meetme.c: This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference. 2007-07-25 22:16 +0000 [r77176] Joshua Colp * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used. 2007-07-25 21:52 +0000 [r77154] Mark Michelson * main/channel.c: chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string 2007-07-25 20:23 +0000 [r77116-77136] Jason Parker * /: so are my fingers... * /: autotagexternals script is still obviously misbehaving... * /: use autotagged externals 2007-07-25 17:14 +0000 [r77071] Joshua Colp * configure, acinclude.m4: Fix autoconf logic for finding OpenH323 when it is not in the first place searched (/usr/share/openh323). 2007-07-25 09:34 +0000 [r77022] Luigi Rizzo * main/rtp.c: set the sequence number in a frame for all frame types 2007-07-25 00:18 +0000 [r76983] Steve Murphy * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 line this fixes bug 10293, where the error message because defaultzone or loadzone was not defined was confusing ........ 2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 Jul 2007) | 2 lines Oops, res contains the error code, not errno. I was wondering why a mutex was reporting "No such file or directory"... ........ * main/app.c: Found another place where we should be using the umask (thanks jcmoore) 2007-07-24 Jason Parker * Asterisk 1.4.9 released. 2007-07-24 16:42 +0000 [r76803-76805] Jason Parker * /: Blocked revisions 76802 via svnmerge ........ r76802 | qwell | 2007-07-24 11:32:04 -0500 (Tue, 24 Jul 2007) | 3 lines Don't create the Asterisk channel until we are starting the PBX on it. (ASA-2007-018) ........ * channels/chan_iax2.c: Don't create the Asterisk channel until we are starting the PBX on it. (ASA-2007-018) 2007-07-24 16:26 +0000 [r76801] Mark Michelson * apps/app_queue.c: Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue. This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone. As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through the member list to figure out how many members there are. Special thanks to blitzrage for helping to test this out. (closes issue #10127, reported by bcnit, patched by me, tested by blitzrage) 2007-07-23 22:38 +0000 [r76708] Tilghman Lesher * apps/app_voicemail.c: It was our stated intention for 1.4 that files created in app_voicemail should depend upon the umask. Unfortunately, mkstemp() creates files with mode 0600, regardless of the umask. This corrects that deficiency. 2007-07-23 18:59 +0000 [r76656] Jason Parker * channels/chan_skinny.c: Fix some incorrect softkey labels in messages. Don't try to play dialtone in some unimplemented features. 2007-07-23 18:29 +0000 [r76654] Joshua Colp * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues. ........ 2007-07-23 17:57 +0000 [r76620] Jason Parker * channels/chan_skinny.c: Don't try to queue up hold/unhold frames on a non-existent channel. Issue 10276. 2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux by putting the value of the digit char in an int. * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines (closes issue #10236) Reported by: homesick Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) Accept Remote Party ID on guest calls. ........ * channels/chan_skinny.c: (closes issue #10268) Reported by: mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak (license 7) Add another OS that has to use the Macros for byte ordering. 2007-07-23 12:25 +0000 [r76485] Russell Bryant * channels/chan_iax2.c: Use a signed integer for storing the number of bytes in the packet read from the network. Using an unsigned value here made it impossible to handle an error returned from recvfrom(). Furthermore, in the case that recvfrom() did return an error, this would cause a crash due to a heap overflow. (closes issue #10265, reported by and fix suggested by timrobbins) 2007-07-22 21:42 +0000 [r76410] Tilghman Lesher * /: Blocked revisions 76409 via svnmerge ........ r76409 | tilghman | 2007-07-22 16:39:55 -0500 (Sun, 22 Jul 2007) | 2 lines We should not use C++ reserved words in API headers (closes issue #10266) ........ 2007-07-21 02:02 +0000 [r76227] Russell Bryant * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) | 4 lines Backport a fix for a memory leak that was fixed in trunk in reivision 76221 by rizzo. The memory used for the localaddr list was not freed during a configuration reload. ........ 2007-07-20 21:36 +0000 [r76211] Steve Murphy * sounds/Makefile: This patch from 10249 is worth applying! It prevents downloading sound files if they are already downloaded. Darn Practical, if you ask me 2007-07-20 21:03 +0000 [r76174-76178] Jason Parker * channels/chan_skinny.c: Allow getting a call from an existing "sub" channel. Cancel ringing if endpoint hangs up before answering. Fixes were backported from trunk (there was apparently a bit of confusion during merge of a previous patch). (closes issue #10241) * main/manager.c: Eliminate a compiler warning with gcc 4.2 by constifying a char * * channels/chan_skinny.c: It's possible for sub->owner to be NULL here if you cancel the call immediately after/during sending a digit. 2007-07-20 18:42 +0000 [r76139] Mark Michelson * apps/app_directory.c: When using users.conf for the entries in the directory, if multiple users had the same last name, only the first user listed would be available in the directory. (closes issue #10200, reported by mrskippy, patched by me) 2007-07-20 18:22 +0000 [r76132] Russell Bryant * main/channel.c: Use the define that specifies the default length of an artificially created DTMF digit in the ast_senddigit() function. The define is set to 100ms by default, which is the same thing that this function was using. But, using the define lets changes take effect in this case, as well as the others where it was already used. 2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 lines (closes issue #10247) Reported by: fkasumovic Patches: chan_sip.patch uploaded by fkasumovic (license #101) Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer. ........ * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic Patches: res_conver.patch uploaded by fkasumovic (license #101) Use the last occurance of . to find the extension, not the first occurance. * apps/app_queue.c: Move makeannouncement variable declaration to proper place. 2007-07-19 20:36 +0000 [r75980] Jason Parker * channels/chan_skinny.c: Remove some duplicate code. 2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson * apps/app_queue.c: The diff on this looks pretty big but all I did was remove a pointless if statement (always evaluates true). * apps/app_queue.c: Changes in handling return values of several functions in app_queue. This all started as a fix for issue #10008 but now includes all of the following changes: 1. Simplifying the code to handle positive return values from ast API calls. 2. Removing the background_file function. 3. The fix for issue #10008 (closes issue #10008, reported and patched by dimas) 2007-07-19 15:53 +0000 [r75928] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) | 6 lines When processing full frames, take sequence number wraparound into account when deciding whether or not we need to request retransmissions by sending a VNAK. This code could cause VNAKs to be sent erroneously in some cases, and to not be sent in other cases when it should have been. (closes issue #10237, reported and patched by mihai) ........ 2007-07-18 22:59 +0000 [r75807] Jason Parker * channels/chan_skinny.c: Need to make sure we set milliseconds and timestamp - pointed out by the recent ast_ time stuff from Tilghman 2007-07-18 21:09 +0000 [r75759] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | 5 lines When traversing the queue of frames for possible retransmission after receiving a VNAK, handle sequence number wraparound so that all frames that should be retransmitted actually do get retransmitted. (issue #10227, reported and patched by mihai) ........ 2007-07-18 20:40 +0000 [r75749] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007) | 2 lines Store prior to copy (closes issue #10193) ........ 2007-07-18 20:17 +0000 [r75732] Jason Parker * channels/chan_skinny.c: Umm, why are we transmitting dialtone on cfwdall? 2007-07-18 20:00 +0000 [r75712] Joshua Colp * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c, pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these Asterisk won't build under devmode using GCC 4.2. 2007-07-18 19:54 +0000 [r75707-75711] Jason Parker * channels/chan_skinny.c: Fixes for 7935/7936 conference phones. Issue 9245, patch by slimey. * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue 9887, patches by DEA 2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007) | 1 line removed the word 'pissed' from ast_log(...) function call for BE-90 ........ 2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp * channels/chan_sip.c: Few more places that needs to check for onhold state. * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless. * channels/chan_h323.c: Don't bother reloading chan_h323 if it did not load successfully in the first place. This would otherwise cause a crash. * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record the threadid of each running thread before shutting them down as the thread themselves may change the value. 2007-07-18 12:29 +0000 [r75529] Tilghman Lesher * apps/app_meetme.c: Using a freed frame causes crashes (closes issue #9317) 2007-07-17 Russell Bryant * Asterisk 1.4.8 released. 2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 Jul 2007) | 3 lines Properly check for the length in the skinny packet to prevent an invalid memcpy. (ASA-2007-016) ........ * main/rtp.c: cast arguments to ast_log so that it builds without warnings for me * channels/iax2-parser.c, channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions 75444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | 5 lines Ensure that when encoding the contents of an ast_frame into an iax_frame, that the size of the destination buffer is known in the iax_frame so that code won't write past the end of the allocated buffer when sending outgoing frames. (ASA-2007-014) ........ * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | 4 lines After parsing information elements in IAX frames, set the data length to zero, so that code later on does not think it has data to copy. (ASA-2007-015) ........ 2007-07-17 20:40 +0000 [r75439] Joshua Colp * main/rtp.c: Ensure that the pointer to STUN data does not go to unaccessible memory. (ASA-2007-017) 2007-07-17 20:33 +0000 [r75437] Russell Bryant * res/res_agi.c: (issue #10210) Reported by: juggie Patches: 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by: juggie, blitzrage Log a warning if someone uses DeadAGI on a live channel. 2007-07-17 20:03 +0000 [r75405] Mark Michelson * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists can return -1 on failure, so I need to be sure that we only enter the if statement if it is successful. Related to my fix to issue #10186 2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant * main/pbx.c: (closes issue #10209) Reported by: juggie Patches: 10209-trunk-2.patch uploaded by juggie Tested by: juggie, blitzrage In ast_pbx_run(), mark a channel as hung up after an application returned -1, or when it runs out of extensions to execute. This is so that code can detect that this channel has been hung up for things like making sure DeadAGI is used on actual dead channels, and is beneficial for other things, like making sure someone doesn't try to start spying on a channel that is about to go away. * res/res_agi.c: Remove a duplicated newline character in AGI debug output. (closes issue #10207, patch by seanbright) 2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming * main/dns.c, /: Merged revisions 75304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007) | 3 lines provide proper copyright/license attribution for this structure that was copied from a BSD-licensed header file long, long ago... ........ * /: another fix that is not needed here (finishing up 75251) 2007-07-16 18:16 +0000 [r75253] Mark Michelson * apps/app_dial.c: Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified. This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up). If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will still continue. (closes issue #10186, reported by jon, patched by me) 2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming * /: block change that is not relevant here 2007-07-13 20:36 +0000 [r75108] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13 Jul 2007) | 3 lines Fix a couple potential minor memory leaks. load_moh_classes() could return without destroying the loaded configuration. ........ 2007-07-13 20:15 +0000 [r75078] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul 2007) | 5 lines Fixed an issue where chanspy flags were uninitialized if no options were passed. What triggered this investigation was an IRC chat where some people's quiet flags were set while others' weren't even though none of them had specified the q option. ........ 2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 Jul 2007) | 6 lines Ensure that adding a user to the list of users of a specific music on hold class is not done at the same time as any of the other operations on this list to prevent list corruption. Using the global moh_data lock for this is not ideal, but it is what is used to protect these lists everywhere else in the module, and I am only changing what is necessary to fix the bug. ........ * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | 12 lines (closes issue #9660) Reported by: mmacvicar Patches submitted by: bbryant, russell Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous When using a TDM400P (and probably other analog cards) there was a chance that you could hang up and pick the phone back up where it has been long enough to be not considered a flash hook, but too soon such that the device reports that it is busy and the person on the phone will only hear silence. This patch makes chan_zap more tolerant of this and gives the device a couple of seconds to succeed so the person on the phone happily gets their dialtone. ........ 2007-07-12 23:00 +0000 [r74998] Mark Michelson * channels/chan_agent.c: Change to my previous fix regarding agent logoff soft. Now uses deferlogoff instead of loginstart since loginstart is used after logoff. Thanks to makoto for pointing this out and suggesting the fix. (closes issue #10178, reported and patched by makoto, with modification by me) 2007-07-12 20:42 +0000 [r74955] Steve Murphy * channels/chan_sip.c: This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up 2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp * main/channel.c: Whoops... didn't want this to be returned to 0 each iteration. * main/channel.c: When waiting for a digit ensure that a begin frame was received with it, not just an end frame. (issue #10084 reported by rushowr) 2007-07-12 16:53 +0000 [r74839-74866] Jason Parker * channels/chan_skinny.c: It helps if I actually add this stuff for the 7921 too - otherwise it won't actually do much of anything. * channels/chan_skinny.c: Add device ID for 7921 wireless skinny phone * channels/chan_skinny.c: Fix dialing in skinny that was broken in some cases. Issue 10136, fix provided by DEA. 2007-07-12 15:53 +0000 [r74815] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul 2007) | 2 lines Only print out a warning for situations where it is actually helpful. (issue #10187 reported by denke) ........ 2007-07-11 22:57 +0000 [r74767] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | 5 lines The function make_trunk() can fail and return -1 instead of a valid new call number. Fix the uses of this function to handle this instead of treating it as the new call number. This would cause a deadlock and memory corruption. (possible cause of issue #9614 and others, patch by me) ........ 2007-07-11 21:14 +0000 [r74722] Mark Michelson * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft" did not work...at all. Now it does. (closes issue #10178, reported and patched by makoto, with slight modification for 1.4 and trunk by me) ........ 2007-07-11 18:34 +0000 [r74657] Russell Bryant * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11 Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows the condition that uses LIKE. This fixes realtime extensions with ODBC. (closes issue #10175, reported by stuarth, patch by me) ........ 2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy * Makefile: This fixes 10172, where the entire man8 dir gets removed during an uninstall of asterisk * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt: further reversion of previously applied floating point stuff for expr2 2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp * /: Blocked revisions 74587 via svnmerge ........ r74587 | file | 2007-07-11 14:15:11 -0300 (Wed, 11 Jul 2007) | 2 lines Use some Makefile magic to determine if linux/compiler.h is present. (issue #10174 reported by francesco_r) ........ * channels/chan_phone.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Instead of figuring out kernel versions that have compiler.h and not... let's just use autoconf to check for it's presence. (issue #10174 reported by francesco_r) * channels/chan_phone.c: Only check if we need to do a SIGMA based tone generation if we have a card. (issue #10179 reported by mikowhy) 2007-07-10 23:32 +0000 [r74476] Mark Michelson * apps/app_voicemail.c: Forwarding a message with IMAP storage was storing the message in the sender's box instead of the forwarded mailbox. (closes issue #10138, reported and patched by jaroth) 2007-07-10 19:58 +0000 [r74374-74428] Jason Parker * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 lines Fix an issue where it was possible to have a service level of over 100% Between the time recalc_holdtime and update_queue was called, it was possible that the call could have been hungup. Move both additions to the same place, so this won't happen. Issue 10158, initial patch by makoto, modified by me. ........ * main/dns.c: Don't use #if to check if something is defined - use #ifdef instead. Pointed out by kpfleming * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4 lines Fix an issue with wrapuptime not working when using AgentLogin. Issue 10169, patch by makoto, with a minor mod by me to not re-break issue 9618 ........ * main/dns.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 74373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines Use res_ndestroy on systems that have it. Otherwise, use res_nclose. This prevents a memleak on NetBSD - and possibly others. Issue 10133, patch by me, reported and tested by scw ........ 2007-07-10 Russell Bryant * Asterisk 1.4.7.1 released. 2007-07-10 16:00 +0000 [r74323] Russell Bryant * res/res_musiconhold.c: fix an uninitialized variable 2007-07-10 15:38 +0000 [r74317] Jason Parker * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 lines Fix a small typo in description in of Voicemail() application. Issue 10170, patch by casper. ........ 2007-07-10 15:31 +0000 [r74314] Russell Bryant * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue #10075, this part reported by jmls on IRC, patch by me) ........ 2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp * /, main/app.c: Merged revisions 74264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 lines Ensure the group information category exists before trying to do a string comparison with it. (issue #10171 reported by mlegas) ........ * channels/chan_sip.c: Only spit out an inringing warning message when it is applicable. Since call limits are already toast in realtime let's not scare the user if they are using it. (issue #10166 reported by bcnit) 2007-07-09 Russell Bryant * Asterisk 1.4.7 released. 2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant * configure, configure.ac: Update the configure script to check for a required function that is not present in the 1.2 version of libpri. This will prevent the configure script from thinking that it has compatible libpri support for Asterisk 1.4, when it actually does not because the installed version is from 1.2. * /: Blocked revisions 74165 via svnmerge ........ r74165 | russell | 2007-07-09 16:00:17 -0500 (Mon, 09 Jul 2007) | 4 lines When the specified class isn't found, properly fall back to the channel's music class or the default. (issue #10123, reported by blitzrage, patches from juggie, qwell, and me) ........ * res/res_musiconhold.c: (closes issue #10123) Reported by: blitzrage Patches submitted by: juggie, qwell, me Tested by: blitzrage When trying to find a music on hold class to use, try all of the options, instead of only the first one that is set. Also, change the MusicOnHold applications to not hang up on the channel when a class can not be found. 2007-07-09 20:19 +0000 [r74159] Jason Parker * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 lines Several chan_zap options were not working on reload because they were arbitrarily disallowed when reloading some/most PRI options (such as signalling) was disallowed. Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload. This corrects that behavior. Issue 9186, patch by tzafrir. ........ 2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson * apps/app_queue.c: Forgot to get rid of an extraneous debug message. * apps/app_queue.c: The n option for Queue should make the queue exit immediately after failure to reach any members and should not be dependent on the timeout value passed to Queue (closes issue #10127, reported by bcnit, repaired by me) 2007-07-09 15:32 +0000 [r74082] Joshua Colp * channels/chan_skinny.c: Only destroy the scheduler context if it was allocated. (issue #10124 reported by gzero) 2007-07-09 14:57 +0000 [r74047] Mark Michelson * apps/app_voicemail.c: Fixed a logic error in leave_voicemail. Pass the mailbox instead of the context to inbox_count when the context is "default." (closes issue #10135, reported by yannj, repaired by me) 2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread synchronization tweaks. (issue #10124 reported by gzero) * configure, acinclude.m4: Use AC_CHECK_HEADER to check for ptlib/openh323 to allow for cross compiling. (issue #9675 reported by zandbelt) 2007-07-09 04:03 +0000 [r73985] Tilghman Lesher * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while 'make progdocs'. (Closes issue #10104) 2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp * main/cdr.c: Give Agent channel names priority when doing CDR merging. (issue #10011 reported by krtorio) * pbx/pbx_config.c: Add a few sanity checks when writing out the dialplan. (issue #10157 reported by dome) 2007-07-08 09:47 +0000 [r73849] Olle Johansson * channels/chan_sip.c: While tracking down a bug, I need some more history. Dumphistory is very useful, indeed. 2007-07-06 23:02 +0000 [r73769] Russell Bryant * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines If a sip_pvt struct has already registered an extension state callback, remove the old one before adding a new one. If this isn't done, Asterisk will crash. (issue #10120) ........ 2007-07-06 16:36 +0000 [r73727] Mark Michelson * apps/app_voicemail.c: Fixing a rare case which causes voicemail to crash when compiled with IMAP storage. inboxcount has the possibility of finding an "interactive" vm_state when no persistent "non-interactive" vm_state exists for that mailbox. If this should happen when someone attempts to leave a message, it results in a crash. This patch, along with my commit in revision 72670 fix issue 10053, reported by jaroth. closes issue #10053 2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras Patches submitted by: Corydon76 Tested by: apsaras Fix a problem with MSSQL 2005 by explicitly stating that '\' is being used as an escape character. ........ * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines (closes issue #10125) Reported by: makoto Patches submitted by: makoto This fixes a crash in chan_sip that happens when the bindaddr setting is not valid on Asterisk startup, gets fixed, and then a reload gets issued. ........ 2007-07-06 15:27 +0000 [r73675] Mark Michelson * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. (issue 9618, reported by jiddings, patched by moi) closes issue #9618 ........ 2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant * BUGS: fix a little spelling error * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop the monitor thread if it was never started. (closes issue #10124, reported by gzero, fixed by me) * channels/chan_iax2.c: copy from the correct buffer when deferring a full frame (related to issue #9937) * channels/chan_iax2.c: * Store the call number that a thread is processing without the full frame bit set to ease debugging * When deferring a full frame for processing, stick it into the queue for the thread that is processing frames for that call, not the one that read the current frame and is about to go back into the idle list (related to issue #9937) 2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729 ........ 2007-07-05 20:50 +0000 [r73512] Russell Bryant * res/res_features.c: Pass HOLD and UNHOLD frames to the other channel when they are returned from a native bridge function. This fixes a problem where when two zap channels are natively bridged and one does a flash hook, the other channel did not receive music on hold. (Reported to me directly by Doug Bailey at Digium) 2007-07-05 19:18 +0000 [r73467] Joshua Colp * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique) ........ 2007-07-05 15:59 +0000 [r73400] Mark Michelson * apps/app_queue.c: Correcting a minor CLI bug I found. When issuing the queue show command, if you type queue show and then press tab, you can continue pressing tab and it will keep auto-completing queue names even though only 1 queue can be used as an argument. 2007-07-05 15:28 +0000 [r73398] Russell Bryant * channels/chan_vpb.cc, channels/Makefile: Make this module build for me in dev-mode 2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 lines Tweak spy locking. (issue #9951 reported by welles) ........ * channels/chan_local.c, /: Merged revisions 73318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 lines Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto) ........ * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 lines Reset ServicelevelPerf variable back to 0 if we are unable to calculate it each time... otherwise we will get previous values. (issue #10117 reported by noriyuki) ........ 2007-07-04 14:53 +0000 [r73208-73253] Christian Richter * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 Jul 2007) | 1 line bchannel configurations like echocancel and volume control, need to be setuped on inbound calls too. ........ * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 Jul 2007) | 1 line bad bug in overlapdial case, we called start_pbx multiple times, because the state wasn't changed.. ........ 2007-07-03 20:17 +0000 [r73143] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing expr floating patch from 1.4; too much of a behavior change. If you want this fix, try trunk instead. bug 9508. 2007-07-03 15:42 +0000 [r73104-73106] Jason Parker * /: What the heck. This should not have happened. * /: use autotagged externals 2007-07-03 12:38 +0000 [r73053] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it does not, because atoi does not distinguish between 0 and error (closes issue #10106) ........ 2007-07-03 08:17 +0000 [r73005] Christian Richter * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only be called from mISDN Source channels.. #9449 ........ 2007-07-02 20:16 +0000 [r72933] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput, main/Makefile, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary. 2007-07-02 18:18 +0000 [r72926] Russell Bryant * main/manager.c: Remove a bogus comment and add proper locking to the handler function for the CLI command to show information on manager actions. 2007-07-02 17:59 +0000 [r72925] Jason Parker * /: Blocked revisions 72924 via svnmerge ........ r72924 | qwell | 2007-07-02 12:58:25 -0500 (Mon, 02 Jul 2007) | 4 lines Fix an issue with playing "oclock" multiple times in French with 24 hour time format. Issue 10101 ........ 2007-07-02 14:32 +0000 [r72888] Joshua Colp * main/channel.c: Added additional DTMF debug messages for when emulation occurs. 2007-07-02 08:41 +0000 [r72850-72852] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 72585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | 1 line check if the bchannel stack id is already used, if so don't use it a second time. Also added a release_chan lock, so that the same chan_list object cannot be freed twice. chan_misdn does not crash anymore on heavy load with these changes. ........ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 72099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | 1 line simplified generation for dummy bchannels, also we mark them as dummies, so they are not used later as real-bchannels, optimized the RESTART mechanisms, we block a channel now on cause:44, and send out a RESTART automatically, then on reception of RESTART_ACKNOWLEDGE we unblock the channel again. ........ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged revisions 72087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | 1 line simplified channel finding and locking a lot. removed unnecessary #ifdefed areas. ........ 2007-07-01 23:52 +0000 [r72806] Russell Bryant * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) | 5 lines When appending lines to call files to keep track of retries, write a leading newline just in case the original call file did not have a newline at the end. This fix is in response to a problem I saw reported on the asterisk-users mailing list. ........ 2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant * configure, configure.ac: Tweak the configure script so that error output isn't spewed to the console when searching for GTK2 libs, and they aren't found. * formats/format_pcm.c: give format_pcm a more concise destription 2007-06-29 19:07 +0000 [r72665] Luigi Rizzo * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for absence of the function. This was already done in trunk. 2007-06-29 Russell Bryant * Asterisk 1.4.6 released. 2007-06-29 16:31 +0000 [r72630] Russell Bryant * /: Blocked revisions 72629 via svnmerge ........ r72629 | russell | 2007-06-29 11:30:56 -0500 (Fri, 29 Jun 2007) | 4 lines Backport changes that make chan_iax2 not start the PBX on an incoming channel until the three-way call setup is completed. These changes are already in 1.4 and trunk. ........ 2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp * main/cdr.c: Minor change for older GCC versions. * Makefile, configure, configure.ac, makeopts.in: Backport fix for GCC versions without support for declaration-after-statement. 2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher * main/manager.c: Issue 10055 - Change memory allocation to use the heap for a command, since the output has the potential to overflow the stack (as it did here) * res/res_jabber.c: Fix 1.4 breakage 2007-06-28 19:44 +0000 [r72493] Russell Bryant * configure, include/asterisk/autoconfig.h.in: regenerate the configure script for rizzo 2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo * configure.ac: add a check for gethostbyname_r so we can simplify the handling e.g. in utils.c Also add comments on a couple of features which are not working on FreeBSD. All the above has been already done in trunk so the merge must be blocked. Can someone please regenerate ./configure ? * Makefile, channels/chan_zap.c, main/say.c: Add -Wdeclaration-after-statement to AST_DEVMODE flags to catch variable declarations in the middle of a block. Fix the few instances of the above spotted out by the compiler. All of this has been already done or is not applicable in trunk, so the merge of this change will be blocked. * apps/app_meetme.c: cast a time_t so that it does not conflict with the print format. This change was already done on trunk so this change needs to be blocked from merging. 2007-06-27 23:29 +0000 [r72383] Brett Bryant * main/asterisk.c, /: Merged revisions 72373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | 3 lines Reinstating patch. This actually fixes the problem, however I was running a development branch without it and mistakenly thought it wasn't fixed. Fixes issue #10010, and #9654: 100% CPU usage caused by an asterisk console losing it's controlling terminal. ........ 2007-06-27 23:25 +0000 [r72381] Joshua Colp * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 lines Update documentation to clarify variable usage with MixMonitor. (issue #9494 reported by netoguy) ........ 2007-06-27 23:03 +0000 [r72335] Brett Bryant * main/asterisk.c, /: Merged revisions 72333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) | 2 lines Reverted changes for earlier revisions 72259 to 72261. Issue #9654, #10010 ........ 2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot) * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 lines Fix issue where queue log events might be missing. (issue #7765 reported by mtryfoss) ........ 2007-06-27 21:08 +0000 [r72272] Russell Bryant * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) | 5 lines Fix a minor issue with parsing the priority number. You could have as much whitespace as you want around a numeric priority, but you couldn't have any whitespace around a special priority like "n" or "hint". (issue #10039, reported by mitheloc, fixed by me) ........ 2007-06-27 20:46 +0000 [r72260] Brett Bryant * main/asterisk.c, /: Merged revisions 72259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) | 4 lines Fixes 100% load when controlling terminal disappears. Issue #9654, #10010 ........ 2007-06-27 20:25 +0000 [r72257] Joshua Colp * main/channel.c, /: Merged revisions 72256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching. ........ 2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming * channels/chan_zap.c: use the proper type for storing group number bits so that if someone specifies 'group=42' it will actually work instead of being silently ignored 2007-06-27 18:40 +0000 [r72182-72185] Jason Parker * /: Blocked revisions 72184 via svnmerge ........ r72184 | qwell | 2007-06-27 13:40:15 -0500 (Wed, 27 Jun 2007) | 4 lines Fix another problem in voicemail with missing symbols. Issue 10074, patch by kryptolus, extended to include #if 0'd blocks (just in case) ........ * apps/app_voicemail.c: Fix another problem in voicemail with missing symbols. Issue 10074, patch by kryptolus, extended to include #if 0'd blocks (just in case) 2007-06-27 17:31 +0000 [r72148] Joshua Colp * main/channel.c: Make the ast_read_noaudio API call behave better under circumstances where DTMF emulation was happening and a generator was setup. (issue #10065 reported by stevefeinstein) 2007-06-27 17:10 +0000 [r72125] Jason Parker * channels/chan_gtalk.c: Don't modify a variable that we don't want modified. Make a copy of it instead. Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts). 2007-06-27 16:34 +0000 [r72112] Russell Bryant * main/rtp.c: Only output debug information related to RTCP timestamps when RTCP debug is turned on (issue #10066, patch by me) 2007-06-27 07:58 +0000 [r72042] Christian Richter * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | 1 line for inbound TE calls, we setup the bchannel when we get the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. removed some #if 0 areas which weren't used anymore. ........ r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | 1 line isdn_lib.c didn't compile ........ 2007-06-27 00:58 +0000 [r72006] Joshua Colp * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work. 2007-06-26 23:02 +0000 [r71953] Mark Michelson * apps/app_voicemail.c: Removing a pointless line. This variable was already set earlier and between then and this line, there is no way that the values on the right side of the assignment could have changed. 2007-06-26 20:36 +0000 [r71915] Jason Parker * main/rtp.c: Don't dereference a pointer that may be NULL here. Issue 10017. 2007-06-26 19:00 +0000 [r71877] Mark Michelson * apps/app_voicemail.c: A few changes, the ultimate goal of which is to keep better track of the number of messages that a mailbox currently has. A description of the changes: 1. Changed the "updated" field of the vm_state struct to act more as a binary semaphore than a counting semaphore, since its current implementation made the inboxcount function not work properly. This change falls in line with a change made by UPenn with their IMAP setup and helps to sync our changes with theirs. 2. Eliminated some redundant calls to get_vm_state_by_mailbox inside leave_voicemail 3. Use the play_folder variable to keep track of the number of old and new messages in a mailbox as the messages are deleted 4. Added an increment to the number of new messages that was not there previously in the leave_voicemail function 2007-06-26 17:49 +0000 [r71848] Jason Parker * /: Blocked revisions 71847 via svnmerge ........ r71847 | qwell | 2007-06-26 12:49:14 -0500 (Tue, 26 Jun 2007) | 4 lines Don't try to install an init script that doesn't exist. Reported to me on #asterisk on Freenode IRC. ........ 2007-06-26 15:47 +0000 [r71796] Mark Michelson * apps/app_voicemail.c: Fixing bug where the authuser was mistakenly pulled from the mailbox string instead of the IMAP user. (closes issue 10054, reported and patched by jaroth) 2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007) | 2 lines Issue 10062 - Trying to move a message without selecting one first results in memory corruption ........ * /, res/res_agi.c: Merged revisions 71656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007) | 2 lines Issue 10035 - handle_exec returns a result inconsistent with all of the other AGI commands ........ 2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp * channels/chan_h323.c: Build a peer as well when hash323 is enabled in users.conf (issue #9599 reported by asagage) * channels/chan_agent.c: Minor tweak for queueing up the unhold frame... this will teach me to do bugs while half asleep. (issue #10046 reported by dimas) 2007-06-25 12:40 +0000 [r71519] Russell Bryant * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue #10048, Matti) 2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick) ........ * main/cdr.c: Fix it so 1.4 actually compiles on my box. * channels/chan_agent.c: Check to make sure the channel pointer is present before queueing up an unhold frame on it. (issue #10046 reported by dimas) 2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant * build_tools/prep_tarball: Include the menuselect-tree file in tarballs to make builds from tarballs a little bit faster * main/asterisk.c, /: Merged revisions 71358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) | 2 lines Revert the patch from issue 9654 due to an unexpected side effect ........ 2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so peer->cdr is what we really wanted to use * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to be able to set variables to the empty string. ........ 2007-06-23 03:29 +0000 [r71230] Steve Murphy * main/cdr.c, res/res_features.c: This patch is meant to fix 8433; where clid and src are lost via bridging. 2007-06-22 22:44 +0000 [r71214] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 Jun 2007) | 1 line fixed a bug that was introduced by copy and paste in the last commit ..bchannels weren't cleaned properly. ........ 2007-06-22 16:05 +0000 [r71128] Joshua Colp * /: Blocked revisions 71124 via svnmerge ........ r71124 | file | 2007-06-22 12:02:40 -0400 (Fri, 22 Jun 2007) | 2 lines Send an unhold indication when going off hold. (issue #10036 reported by speedy) ........ 2007-06-22 15:38 +0000 [r71096-71123] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 70672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | 1 line we activate the bchannels in TE mode on incoming calls only when we want to connect the call. ........ * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 Jun 2007) | 1 line forgot one place .. ........ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 Jun 2007) | 1 line on receiption of cause:44 we mark the channel as in use and inform the user about the situation, we need to test the RESTART stuff then. Also shuffled the empty_chan_in_stack function after the bchannel cleaning functions, to avoid race conditions. ........ * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 Jun 2007) | 1 line when we send out a SETUP, but get no response, we should cleanup everything after reception of a hangup. ........ * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | 1 line restart indicator 0x80 is correct, at least that's what libpri does. ........ * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 Jun 2007) | 1 line if the bridged partner is mISDN too we should not send dtmf tones, they are transmitted inband always ........ * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 Jun 2007) | 1 line if we have already some digits, we just stop the tones. ........ 2007-06-22 15:00 +0000 [r71068] Jason Parker * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged revisions 71065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 lines Fix a few silly usages of ast_playstream() - it only ever returns 0... Issue 10035 ........ 2007-06-22 14:53 +0000 [r71066] Brett Bryant * main/asterisk.c, /: Merged revisions 71064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | 10 lines Fixed infinite loop when controlling terminal was lost and return value of input function wasn't checked for errors. This would cause 100% cpu to be taken up. (closes issue #9654, issue #10010) Reported by: mnicholson, and eserra Idea for the patch from mnicholson, patched by me ........ 2007-06-22 14:10 +0000 [r71063] Steve Murphy * main/cdr.c: My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone. 2007-06-22 03:14 +0000 [r71003] Russell Bryant * channels/chan_iax2.c: Fix a small typo which ... well ... completely broke chan_iax2. oops! (issue #9937, patch by me) 2007-06-21 22:34 +0000 [r70949] Steve Murphy * main/cdr.c, /: Merged revisions 70948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 line This little fix is in response to bug 10016, but may not cure it. The code is wrong, clearly. In a situation where you set the CDR's amaflags, and then ForkCDR, and then set the new CDR's amaflags to some other value, you will see that all CDRs have had their amaflags changed. This is not good. So I fixed it. ........ 2007-06-21 21:40 +0000 [r70899] Joshua Colp * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 lines Don't explode if the gain option is specified without a value. (issue #9274 reported by mfarver) ........ 2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant * channels/chan_iax2.c: Put the thread reading from the socket back in the idle list if it deferred the processing of a full frame to another thread * channels/chan_iax2.c: If a full frame is received while one of the iax2 threads is in the middle of handling a full frame for the same call, queue it up for processing by that same thread later instead of dropping it. (issue #9937, patch by me) 2007-06-21 20:19 +0000 [r70841] Steve Murphy * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 line it was pointed out that the cdr_custom config load could get a lock, and under certain circumstances, would never release it. I also noted that the situation where more than one mapping spec was warned about, but did not ignore further mappings as it had promised. I think I have fixed both situations. ........ 2007-06-21 19:49 +0000 [r70808] Mark Michelson * apps/app_voicemail.c: When volgain is used don't leave a temporary file behind. (Closes Issue 8514, Reported and patched by ulogic, code reviewed by Jason Parker) 2007-06-21 15:22 +0000 [r70727] Joshua Colp * main/rtp.c: Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan) 2007-06-21 15:21 +0000 [r70726] Russell Bryant * apps/app_meetme.c: Remove a couple of duplicate unlocks 2007-06-21 13:58 +0000 [r70677] Joshua Colp * apps/app_voicemail.c: Fix building with ODBC storage enabled. (issue #10025 reported by denisgalvao) 2007-06-21 13:00 +0000 [r70656] Steve Murphy * main/cdr.c: Via complaints aired in asterisk-users, I submit these changes, which allow cdr updates to see macro context/exten, whether hung up or not 2007-06-20 23:32 +0000 [r70554-70612] Jason Parker * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql. Issue 10020, patch by my, with credit to prashant_jois for pointing out the problem. * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race condition fix * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur when reloading the module. Issue 10022, patch by me, with credit to prashant_jois for finding the bug. 2007-06-20 22:22 +0000 [r70552] Joshua Colp * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith) ........ 2007-06-20 20:53 +0000 [r70494] Jason Parker * channels/chan_skinny.c: Make sure we clear the previously dialed number if it did not exist. Issue 9958. 2007-06-20 19:29 +0000 [r70445] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) | 2 lines Issue 9997 - Timelimit times out the wrong channel ........ 2007-06-20 18:46 +0000 [r70397] Russell Bryant * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | 5 lines Fix a problem where an established call would not be properly disconnected when a PRI disconnect is received depending on which cause code was received. (issue #9588, original patch by softins, updated patch from jtexter3, and some additional feedback from mhardeman) ........ 2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp * main/rtp.c, main/frame.c: Put the speex packetization values back in but disable it when setting up the smoother. * main/frame.c: Don't do packetization/smoother stuff with speex, it doesn't work. 2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant * contrib/scripts/ast_grab_core: don't delete the backtrace in ast_grab_core * channels/chan_gtalk.c: Only attempt to queue a hangup on the owner channel if it actually exists. (issue #9795, patch from zandbelt) 2007-06-19 18:23 +0000 [r70062] Steve Murphy * main/channel.c, /: Merged revisions 70053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 line This fixes 9246, where channel variables are not available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to consolidate the channel variables during a masquerade, and then copy the merged variables back onto the clone, so the zombie has the same vars that the 'original' has. ........ 2007-06-19 17:07 +0000 [r70003] Joshua Colp * main/rtp.c, /: Merged revisions 69992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu) ........ 2007-06-19 16:46 +0000 [r69991] Russell Bryant * /: Blocked revisions 69990 via svnmerge ........ r69990 | russell | 2007-06-19 11:45:37 -0500 (Tue, 19 Jun 2007) | 12 lines Backport fix for crashes related to subscriptions from 1.4 ... Fix a crash that could occur when handing device state changes. When the state of a device changes, the device state thread tells the extension state handling code that it changed. Then, the extension state code calls the callback in chan_sip so that it can update subscriptions to that extension. A pointer to a sip_pvt structure is passed to this function as the call which needs a NOTIFY sent. However, there was no locking done to ensure that the pvt struct didn't disappear during this process. (issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use the sip_pvt lock wrappers by eliel) ........ 2007-06-19 16:24 +0000 [r69987] Joshua Colp * main/channel.c, /: Merged revisions 69986 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 lines Update BRIDGEPEER variable if set to the new channel name when a masquerade happens. (issue #9699 reported by dimas) ........ 2007-06-19 15:22 +0000 [r69944] Russell Bryant * channels/chan_sip.c: Fix a crash that could occur when handing device state changes. When the state of a device changes, the device state thread tells the extension state handling code that it changed. Then, the extension state code calls the callback in chan_sip so that it can update subscriptions to that extension. A pointer to a sip_pvt structure is passed to this function as the call which needs a NOTIFY sent. However, there was no locking done to ensure that the pvt struct didn't disappear during this process. (issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use the sip_pvt lock wrappers by eliel) 2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 lines Perform an extra hangup check just in case. (issue #9589 reported by bcnit) ........ * /, res/res_features.c: Merged revisions 69846 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 lines Add parked call extension AFTER the parking slot has been announced, otherwise two threads will try to handle the same channel and it will go kaboom. (issue #9191 reported by japple) ........ * main/callerid.c: Fix for building on PowerPC under Linux. 2007-06-18 19:48 +0000 [r69796] Tilghman Lesher * channels/chan_sip.c: Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels 2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp * channels/chan_sip.c: Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc) * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin) ........ 2007-06-18 17:46 +0000 [r69744] Tilghman Lesher * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007) | 2 lines Issue 9998 - Remove SIG prefix, since it's not supported by ksh ........ 2007-06-18 16:51 +0000 [r69708] Joshua Colp * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called for the first time so that it does not needlessly spit out changed messages when the host really didn't change. 2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c, build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c: To prevent 92138749238754 more reports of "I have unixodbc installed, but still can't build *_odbc.so!", check for ltdl directly, instead of just listing it as another library to include in the unixodbc check in the configure script. This also makes ltdl show up as a dependency in menuselect so people know what to go install. (related to issue #9989, patch by me) * build_tools/prep_moduledeps: Change the use of "echo -e" to "printf". On systems where /bin/sh is not bash, most of the lines in menuselect-tree were getting a "-e" at the beginning of every line. I'm surprised nobody noticed this, but I think the XML parser was being very nice and ignoring them. 2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp * channels/chan_sip.c: Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working. * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls. 2007-06-18 15:46 +0000 [r69660] Russell Bryant * Makefile: Tweak paths for BSD systems (issue #10001, stuarth) 2007-06-18 13:55 +0000 [r69625] Joshua Colp * channels/chan_sip.c: Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi) 2007-06-15 Russell Bryant * Asterisk 1.4.5 released. 2007-06-15 20:18 +0000 [r69579] Russell Bryant * res/res_features.c: Fix a silly deadlock in res_features that I found while debugging on one of blitzrage's test machines. It was one of the situations where he was seeing hung channels, and may be the cause of some of the reports from other people. (related to issue #9235) 2007-06-15 19:23 +0000 [r69558] Joshua Colp * apps/app_speech_utils.c: Add support for setting the maximum length of acceptable DTMF in SpeechBackground. 2007-06-15 15:27 +0000 [r69518] Russell Bryant * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated "SUCCESS" for both an answer of the incoming call on the trunk, or if the trunk reached its ring timeout. This patch changes the variable to say "RINGTIMEOUT" in that case. (issue #9973, reported by n00dle, patch by me) 2007-06-14 23:22 +0000 [r69434-69470] Jason Parker * main/config.c, /: Merged revisions 69469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 lines Fix an issue where the line number in an unterminated comment block error message would show the wrong line number. "Reported" to me on #asterisk (somebody posted an error message, and I happened to catch it) ........ * sounds/Makefile: Update to latest versions of sound files. 2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c, main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c, apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c, main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c: use ast_localtime() in every place localtime_r() was being used 2007-06-14 21:08 +0000 [r69358] Russell Bryant * main/say.c: Fix some problems with saying dates and times for the "tw" langauge (issue #9964, ljmid) 2007-06-14 15:21 +0000 [r69259] Jason Parker * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun 2007) | 4 lines Change a quite broken while loop to a for loop, so "continue;" works as expected instead of eating 99% CPU... Issue 9966, patch by me. ........ 2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp * channels/chan_iax2.c: Whoops... * channels/chan_iax2.c: Let's make chan_iax2 media only native transfers actually work. (issue #9376 reported by simone cittadini) * channels/iax2-parser.c: Add TXMEDIA to list so that it is properly displayed during iax2 packet output. 2007-06-13 19:57 +0000 [r69183] Russell Bryant * channels/chan_sip.c: Move the logic for destroying a call when no response is received to a BYE outside of the block that checks for FLAG_FATAL to be set. This flag is only set when the packet is transmitted with the reliability set to XMIT_CRITICAL when the original packet is transmitted. A BYE is always sent with it set to XMIT_RELIABLE, meaning this code could never be encountered. This resulted in seeing some SIP channels that would never go away with the last packet sent being a BYE. (part of issue #9235, patch from jcmoore) 2007-06-13 19:41 +0000 [r69181] Mark Michelson * apps/app_voicemail.c: Contains a patch for fixing an encoding problem when using Outlook to view voicemail emails and attachments. This fix has also been tested on Thunderbird, Evolution, Pine, and Mutt. (Issue 9336, reported by marwick, patched by mutterc) 2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp * apps/app_meetme.c: Really ignore NULL frames and check whether the channel hungup or not. (issue #9912 reported by junky) * /, main/app.c: Merged revisions 69127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 lines Return group counting to previous behavior where you could only have one group per category. (issue #9711 reported by irroot) ........ 2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant * channels/chan_sip.c: Clarify a bit of logic. This doesn't change behavior in any way, but it is helpful when following the logic to debug problems like 9235. * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct was accessed without the lock held. This issue was reported to me via email by Dmitry Mishchenko. Thanks! * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois in #asterisk-bugs. PQclear() was not called on the result structure after doing a PQexec(). Also, fix up some formatting in passing. 2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp * channels/chan_iax2.c: Change the full frame dropping log message to debug to avoid future bug reports. * channels/chan_iax2.c: Schedule the sending of a PING packet a second later than previously so that it does not collide with the LAGRQ. 2007-06-12 19:13 +0000 [r69010] Russell Bryant * main/channel.c: In ast_channel_make_compatible(), just return if the channels' read and write formats already match up. There are code paths that call this function on a pair of channels multiple times. This made calls fail that were using g729 in some cases. The reason is that codec_g729a will unregister itself from the list of available translators will all licenses are in use. So, the first time the function got called, the right translation path was allocated. However, the second time it got called, the code would not find a translation path to/from g729 and make the call fail, even if the channel actually already had a g729 translation path allocated. (SPD-32) 2007-06-12 14:23 +0000 [r68922] Joshua Colp * main/rtp.c, /: Merged revisions 68921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines Bring RTP back to Asterisk at the end of a native bridge no matter what. ........ 2007-06-11 21:20 +0000 [r68814] Jason Parker * include/asterisk/time.h: Solaris 10 sometimes (?) needs this include in order to have NULL defined. 2007-06-11 20:45 +0000 [r68781] Tilghman Lesher * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly used 2007-06-11 16:57 +0000 [r68733] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 68732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | 1 line added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0 ........ 2007-06-11 14:33 +0000 [r68683] Joshua Colp * main/channel.c, /: Merged revisions 68682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines Improve deadlock handling of the channel list. (issue #8376 reported by one47) ........ 2007-06-11 10:29 +0000 [r68644] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | 1 line fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too. ........ 2007-06-11 04:21 +0000 [r68595] Tilghman Lesher * pbx/pbx_config.c: "dialplan save" produced garbage in the config file 2007-06-08 22:23 +0000 [r68527] Russell Bryant * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | 4 lines Don't automatically hang up after running Dictate so that callers can exit cleanly using '#' (closes issue #9577, patch from Thomas Andrews) ........ 2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming * channels/chan_iax2.c: actually remember the type/subclass of full frames that are in process 2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp * /, main/say.c: Merged revisions 68397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 lines Don't call ast_waitstream_full when the control file descriptor and audio file descriptor are not set, simply call ast_waitstream! (issue #8530 reported by rickead2000) ........ * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 lines Do a DNS lookup immediately upon calling the dnsmgr function, don't wait until a refresh happens. (issue #9097 reported by plack) ........ 2007-06-07 23:14 +0000 [r68354] Russell Bryant * /, main/say.c: Merged revisions 68351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | 3 lines Fix a problem where saying a character wouldn't properly break out when the caller pressed '#' (issue #8113, reported by patbaker82, patch from jamesgolovich (hey, long time no see!) and patbaker82) ........ 2007-06-07 23:00 +0000 [r68326] Jason Parker * apps/app_voicemail.c: Fix incorrect French syntax of "old messages". Request for feedback was sent to asterisk-dev mailing list, with little response. Issue 9118, patch by junky. 2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming * channels/chan_iax2.c: some improvements to the IAX2 full frame dropping logic recently added: - use inaddrcmp(), since we have it - output the type of frame and subclass being dropped, and the type/subclass that is already being processed (which caused the drop) 2007-06-07 21:16 +0000 [r68280] Russell Bryant * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent queue members when using realtime configuration for queues. Also, remove an unneeded leading slash for the astdb family. (issue #9911, patch by atis) 2007-06-07 20:25 +0000 [r68211-68249] Jason Parker * channels/chan_skinny.c: Fix an issue with newer phones which require packets be padded out to the correct length. Issue 9887, patch by DEA. * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 lines Don't try to save voicemail greetings unless the user presses '1' to accept/save. Issue 9904, patch by me. ........ 2007-06-07 19:47 +0000 [r68198] Mark Michelson * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a check to make sure that greetings get stored properly. (Issue 8016, reported by edhorton, patched by alamantia with modification by me. Thanks to Jason Parker for the advice on this). 2007-06-07 19:46 +0000 [r68196] Olle Johansson * channels/chan_features.c: Disable chan_features by default in menuselect 2007-06-07 19:30 +0000 [r68192] Russell Bryant * main/strcompat.c: Include stdarg.h for build issues on Solaris (issue #9381) 2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp * main/channel.c: Fix logic when doing a name based channel search for a structure when you want to start from a specific point in the channel list. (issue #9324 reported by slavon) * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 lines Allow the 'g' option to work if used with the 'S' option. (issue #9888 reported by gasparz) ........ 2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't forget. * res/res_jabber.c: Ok, we found out that this is not about if you have any *active* clients using TLS, but if you have initialized TLS at all during the lifetime of the module. So if you reload to disable TLS, it won't help. * res/res_jabber.c: If you have a jabber client that uses TLS, refuse unload. Bad fix, but will prevent crashes while we are trying to find a workaround. Iksemel development seems to have stalled and we might have to stop using the TCP/TLS connections in that library and use our own, which would scale better from a poll/select perspective I guess. It would also make it easier to migrate to OpenSSL and stop Asterisk from depending on both OpenSSL and GnuTLS. * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make sure we can unload res_jabber. Patch by phsultan - thanks! Due to a bug in the iksemel library, this will not work if you are using GTLS in the connection. That's being investigated. If you figure out a way to handle that without us having to patch iksemel, let us know in the bug report. Thanks. 2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia) ........ * main/file.c: Properly handle cases where a stream can't be written to. (issue #9757 reported by junky) 2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant * res/res_snmp.c: Disable reload functionality in res_snmp. It is not possible to initialize the snmp library more than once without completely unloading the module and loading it again. (issue #9571, reported by hristo, additional helpful debug information from festr, patch from me) * channels/chan_sip.c: Fix a crash when doing call pickups with SIP phones. The code unlocked the channel when it should not have. (issue #9652, reported by corruptor, fixed by me) 2007-06-06 19:26 +0000 [r67804] Mark Michelson * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault under a specific set of circumstances: 1. VoiceMailMain was configured in the dialplan with an extension as its argument 2. A message was left for this mailbox 3. Tried to call VoiceMailMain but hung up before entering password. This was fixed by checking that a pointer was non-null prior to trying to dereference it. (Issue 9810, reported by xmarksthespot, patched by Corydon76 with modifications by me). 2007-06-06 16:55 +0000 [r67716] Russell Bryant * main/channel.c, /, include/asterisk/linkedlists.h: Merged revisions 67715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines We have some bug reports showing crashes due to a double free of a channel. Add a sanity check to ast_channel_free() to make sure we don't go on trying to free a channel that wasn't found in the channel list. (issue #8850, and others...) ........ 2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp * main/rtp.c, /: Merged revisions 67649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz) ........ * main/translate.c: Fix plc_samples warning when registering a translator. (issue #9897 reported by xylome) * apps/app_directed_pickup.c: Include macroexten while searching for a channel to pick up in case they are in a macro. (issue #9491 reported by jamesb63) * res/res_agi.c: Make the new "agi debug off" CLI command work. (issue #9890 reported by eliel) * /, main/devicestate.c: Merged revisions 67593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 lines Revert channel name splitting fix for Zap. The moral of the story is don't use - in your user/peer names. (issue #9668 reported by stevedavies) ........ 2007-06-05 23:01 +0000 [r67558] Russell Bryant * apps/app_meetme.c: Fix some crashes related to the use of the "meetme" CLI command. The code for this command was not locking the conference list at all. (issue #9351, reported by and patch submitted by Junk-Y, committed patch is different and by me) 2007-06-05 21:30 +0000 [r67526] Steve Murphy * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug 9883, wherein macros were not allowing the includes construct. fixed and tested, looks OK. Now includes can serve as an adjunct to catch. 2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant * include/asterisk/linkedlists.h: This bug has been hanging over my head ever since I wrote this SLA code. Every time I tried to go debug it by adding some debug output, the behavior would change. It turns out I wasn't crazy. I had the following piece of code: if (remove) AST_LIST_REMOVE_CURRENT(...); Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional statement didn't do much good at all. It always ran at least all of the macro minus the first statement, so I was seeing list entries magically disappear when they weren't supposed to. After many hours of debugging, I have come to this extremely irritating fix. :) (issues #9581, #9497) * channels/chan_zap.c: Suppress a bunch of debug output unless option_debug is on 2007-06-05 18:32 +0000 [r67424] Mark Michelson * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails saved to IMAP storage using extensions other than gsm were unable to be played over the phone. (Issue 9786, reporter: xmarksthespot, Patched by xmarksthe spot with revisions by me, reviewed by Russell Bryant). 2007-06-05 18:18 +0000 [r67421] Jason Parker * channels/chan_skinny.c: Correctly update date/time on devices throughout the life of the device, instead of just at registration. Issue 9152, yet another patch by DEA. 2007-06-05 18:17 +0000 [r67420] Steve Murphy * pbx/pbx_ael.c: Added code to automatically add a default case to switches that don't have one. In some cases, rather than fall thru, it results in a goto with -1 result, which terminates the extension; a sort of dialplan seqfault, sort of. This was required to fix bug reported in 9881 2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant * main/channel.c: Handle a failure in malloc() in ast_safe_string_alloc() * main/channel.c: Fix a problem that showed itself by causing Zap channel names to be completely bogus on my machine. ast_safe_string_alloc() was broken. It called vsnprintf() on a va_args list twice without re-initializing it. After the first usage, va_end() and va_start() must be called again. 2007-06-05 16:14 +0000 [r67329-67334] Christian Richter * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | 1 line briding is a bool, fixed copy and paste issue. ........ * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination. ........ 2007-06-05 15:51 +0000 [r67308] Russell Bryant * main/asterisk.c, main/loader.c, include/asterisk/module.h: When shutting down "gracefully", go through and run the unload() callbacks for all of the modules. "stop now" is considered a non-graceful shutdown and will not go through this process. (issue #9804, reported by chrisost, patch by me) 2007-06-05 15:22 +0000 [r67304] Joshua Colp * channels/chan_iax2.c: Only muck with the thread structure if an idle one was found/created. 2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming * channels/chan_iax2.c: ensure that a burst of full frames (AST_FRAME_DTMF being the prime example) will not be processed out of order... this is a brute force fix, but seems to be the safest fix for now (thanks to the Digium PQ department for finding this bug) 2007-06-05 10:25 +0000 [r67210] Christian Richter * channels/misdn_config.c, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: Merged revisions 67209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line added possibility to deactivate bridging per port ........ 2007-06-04 23:43 +0000 [r67162] Tilghman Lesher * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007) | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops. ........ 2007-06-04 23:31 +0000 [r67158] Russell Bryant * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt structure can disappear and the code did not account for it and crashes. (issues #9642, #9569, #9666, probably others ... based on the work by stevedavies and mihai, with additional changes from me) 2007-06-04 23:26 +0000 [r67121-67156] Jason Parker * channels/chan_skinny.c: Fix for skinny keepalives. If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily adding 10% for network issues, etc), unregister the device. Issue 8394, patch by DEA. * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny) Issue 9855, patch by DEA. 2007-06-04 22:28 +0000 [r67119] Russell Bryant * channels/chan_iax2.c: Add comments for two functions that get called with the appropriate call locked, but perform operations that could result in the pvt structure getting destroyed before returning again, causing numerous seg faults all over the module. (inspired by issues #9642, #9569, and #9666, and the work done by stevedavies and mihai) 2007-06-04 21:59 +0000 [r67073] Steve Murphy * main/cdr.c: This typo has been here since 1.4 forked. It has been the source of heartburn to many a dialplan/CDR programmer. 2007-06-04 21:47 +0000 [r67071] Russell Bryant * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC) 2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp * channels/chan_sip.c: Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo) * apps/app_dial.c: Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661 reported by dimas) * res/res_features.c: Returning a value that indicates the parking of a call was a success when it really wasn't (because the parking slot selected was in use) is the wrong thing to do. (issue #9723 reported by mdu113) 2007-06-04 17:11 +0000 [r67061] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.mandrake.zaptel, contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) | 2 lines Add revision Id tags (by request of tzafrir) ........ 2007-06-04 16:02 +0000 [r67026] Russell Bryant * configure, configure.ac: Change the configure script to build a test program against libcurl to make sure the results from curl-config can be used to compile successfully. This is intended to help prevent a situation where you are cross compiling, and the configure script finds the curl library installed on the host. (issue #9865, reported and patched by zandbelt) 2007-06-04 15:50 +0000 [r67021] Tilghman Lesher * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash 2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that was marked as the final transmission for a call, don't call iax2_destroy() for that call while the global frame queue is still locked. There is a very nice explanation of the deadlock in the report. (issue #9663, thorough report and patch from stevedavies, additional positive test reports from mihai and joff_oconnell) * include/asterisk/stringfields.h: Fix some compiler warnings in C++ modules. (issue #9866, reported by osk, patch by Corydon76) 2007-06-01 21:45 +0000 [r66919] Tilghman Lesher * funcs/func_odbc.c: On some drivers, deallocating the statement handle isn't enough. We also have to clear the cursor (nice, Oracle) 2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson * apps/app_voicemail.c: Removing extraneous debugging lines from revision 66897. Sorry :) * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP storage. Attachments with format specified as gsm were duplicated (i.e. two attachments) were left. Thank you very much to xmarksthespot for submitting the patch that fixed this. (Issues 9787 and 8873, Reported by xmarksthespot and jerjer, patched by xmarksthespot) 2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant * channels/chan_skinny.c: Changes to the way DTMF is handled in the core broke dialing in chan_skinny. This patch makes chan_skinny usable again. I did not end up testing this, but there are multiple positive test reports listed in the bug report. (issue #9596, reported by pj, testing by pj and mvanbaak, and the fix was written by DEA) * apps/app_page.c: List app_meetme as a module that app_page depends on. 2007-05-31 23:03 +0000 [r66821] Tilghman Lesher * doc/asterisk.8: Issue 9850 - update preferred command line syntax 2007-05-31 18:41 +0000 [r66775] Russell Bryant * res/res_speech.c, include/asterisk/app.h, include/asterisk/speech.h: Change a couple of header files to not use "new", which is a reserved keyword in C++. (issue #9830, reported by osk) 2007-05-31 17:15 +0000 [r66770] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime, but only because we lack core API to do it. ........ 2007-05-31 16:14 +0000 [r66768] Joshua Colp * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx) ........ 2007-05-30 23:26 +0000 [r66671] Mark Michelson * apps/app_voicemail.c: Fixed seg-faults when recording greetings in voicemail with IMAP enabled. (Issue No. 9735, reported by xmarksthespot, patched by me) 2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp * channels/chan_sip.c: Silly me for having out of date source! Oh well... I'm still leaving my comment. * channels/chan_sip.c: When calling some peer/host that may not exist/reply back... don't keep the dialog in memory for all of eternity. * channels/chan_zap.c, channels/chan_features.c: Change how channel names are generated a bit. (issue #9825 reported by eldadran) 2007-05-29 21:56 +0000 [r66538] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) | 2 lines If the value of a variable passed to FIELDQTY is blank, then FIELDQTY should return 0, not 1. ........ 2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson * channels/chan_sip.c: Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response. * channels/chan_sip.c: Don't issue hangup on hangup on hangup on hangup (for jcmoore) 2007-05-29 16:44 +0000 [r66437] Joshua Colp * main/rtp.c: Handle cases where a frame may have no data. (issue #9519 reported by dmb) 2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson * channels/chan_sip.c: Don't reset hangupcause if we already have one * channels/chan_sip.c: Tracking down hanging channels, killing them one by one. Issue #9235 and related 2007-05-29 15:43 +0000 [r66398] Joshua Colp * doc/datastores.txt: Update datastores documentation. (issue #9801 reported by mnicholson) 2007-05-29 09:41 +0000 [r66363] Olle Johansson * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines Issue #9802 - Change inuse counter on CANCEL ........ 2007-05-28 23:16 +0000 [r66312] Joshua Colp * channels/chan_zap.c: Make the usedistinctiveringdetection option work again. (issue #9823 reported by premeau) 2007-05-27 04:12 +0000 [r66244] Jason Parker * channels/chan_zap.c: I don't know what this was trying to do, but it's clearly incorrect. Issues 9808 and 9809. 2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming * configure, configure.ac: have to check for OSP toolkit _after_ checking for OpenSSL 2007-05-25 14:41 +0000 [r66159] Tilghman Lesher * /, main/say.c: Merged revisions 66127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch ........ 2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming * configure, configure.ac, channels/chan_gtalk.c, makeopts.in, res/res_jabber.c: handle the GNUTLS library properly in the configure script and build system don't build in OSP support unless we have found and are allowed to use SSL support 2007-05-24 22:23 +0000 [r66076] Russell Bryant * main/channel.c: if the string field init fails, clean up the stuff that was allocated already 2007-05-24 22:16 +0000 [r66074] Joshua Colp * main/slinfactory.c: Fix slinfactory logic when dealing with frames coming in that may already be in the signed linear format. 2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant * main/channel.c: Check the result of ast_string_field_init() in ast_channel_alloc() * main/rtp.c: Make 1.4 build on my machine, too.. 2007-05-24 20:54 +0000 [r66029-66030] Jason Parker * configure: Rebuild configure script for previous ar fix. * configure.ac: Following moving strip to AC_PATH_TOOL, we need to do something similar for ar. 2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant * configure, include/asterisk/autoconfig.h.in, configure.ac: Checking for the strip application needs to be done with AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross compilation environments. * Makefile: Clear CFLAGS before running make for menuselect. (issue #9784, reported by ovi, patch by me) 2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming * channels/chan_gtalk.c: oops, use #ifdef instead of #if * channels/chan_gtalk.c: don't reference GnuTLS headers and functions unless the configure script found it * main/rtp.c: don't use uninitialized variables 2007-05-24 15:27 +0000 [r65902] Joshua Colp * main/manager.c: Add the ability to blacklist certain commands from being executed using the Command AMI action. (issue #9240 reported by junky) 2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly. * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks! 2007-05-24 15:16 +0000 [r65877-65883] Jason Parker * .cleancount: Update cleancount for that last commit - just for good measure. * include/asterisk/translate.h, codecs/codec_speex.c, main/translate.c, codecs/codec_ilbc.c: Fix handling of zero-length frames when a codec is capable of native PLC. Issue 9183, patch by Mihai. 2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard * funcs/func_math.c: merged qwell's func_math patch for issue 9507 2007-05-24 15:08 +0000 [r65863] Joshua Colp * main/rtp.c: I like it when the RTP stack compiles myself... 2007-05-24 15:05 +0000 [r65857] Olle Johansson * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat. 2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant * apps/app_festival.c: Ensure that frames are fully initialized. This will probably fix getting weird timestamp log messages in logs when using the Festival app. (issue #9781, patch by me) * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous code would result in oscillating and incorrect data. Additionally, the RTT would sometimes report negative values due to incorrect calculations. (issue #9601, patch from davetroy) 2007-05-24 14:48 +0000 [r65841] Olle Johansson * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for jingle 2007-05-24 14:42 +0000 [r65839] Joshua Colp * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford) ........ 2007-05-24 14:38 +0000 [r65836] Olle Johansson * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan - Fix "login" as component to jabber server. ...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits of BYE requests. 2007-05-24 09:37 +0000 [r65768] Christian Richter * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example. ........ 2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming * channels/chan_iax2.c: start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines ensure that variables are set on a newly created channel before we start a PBX on it ........ * channels/chan_iax2.c: clear the 'delay PBX' flag when we are ready to start the PBX * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else) * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it ........ 2007-05-23 13:07 +0000 [r65589] Russell Bryant * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines Revert revision 62417 as someone reported problems with it to Mark. This was related to issue #9588. ........ 2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming * build_tools/make_version: when building a version string for a developer branch, include the base branch in the version string 2007-05-22 18:40 +0000 [r65501] Russell Bryant * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a dependency for app_voicemail and chan_zap (Thanks to mnicholson for pointing it out) 2007-05-22 15:04 +0000 [r65452] Joshua Colp * apps/app_meetme.c: Remove a double const. 2007-05-22 14:02 +0000 [r65408] BJ Weschke * apps/app_followme.c: Fix a problem with flag recognition. 2007-05-22 13:09 +0000 [r65394] Russell Bryant * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | 4 lines Fix a memory leak that I just noticed in the device state handling in app_queue. On most device state changes, it would leak roughly 8 to 64 bytes (the length of the name of the device). ........ 2007-05-22 08:12 +0000 [r65342] Christian Richter * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message ........ 2007-05-20 17:59 +0000 [r65250] Joshua Colp * res/res_agi.c: res_agi needs to export two symbols (ast_agi_register and ast_agi_unregister) for usage by others. (issue #9755 reported by mnicholson) 2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions 65172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before. ........ 2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines Not getting an ACK to a 200 OK in the initial invite is critical to the call. ........ * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no other patch) if you have problems with hanging SIP channels. Thank you. A special Thank You to WeBRainstorm that gave me access to his system. ........ * channels/chan_sip.c: - Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp. - In order to not duplicate code, I restructured some of the code for putting calls on/off hold. Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now. 2007-05-18 12:40 +0000 [r65039] Christian Richter * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged revisions 65007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position. ........ 2007-05-18 10:37 +0000 [r64974] Olle Johansson * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of call 2007-05-18 08:58 +0000 [r64904] Christian Richter * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested. ........ 2007-05-18 02:48 +0000 [r64868] Russell Bryant * apps/app_queue.c: Fix a small bug I noticed while working on something else. app_queue did not unregister its device state monitoring callback in unload_module(). So, this would make Asterisk crash on the first device state change after you unload the module. 2007-05-17 21:19 +0000 [r64820] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 64819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) | 2 lines How is it that we never caught that this is returning the opposite of our documentation, until now? ........ 2007-05-17 16:53 +0000 [r64761] Jason Parker * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 lines If we have a negative current message, we shouldn't go back even further... Issue 9727. ........ 2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant * contrib/scripts/astxs (removed): Remove script that is no longer functional since the build system was redone. (issue #9340, reported by junky) * apps/app_dial.c: Increase the size of a buffer to support longer dial strings for channels. (issue #9291, reported and fix suggested by meni) 2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp * channels/chan_sip.c: Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu) * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported by xmarksthespot) 2007-05-17 06:13 +0000 [r64686] Russell Bryant * README: Update the main README to reflect the new build process for 1.4 and above. (issue #9725, patch by eliel) 2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson * /: Blocking patch already in this code * channels/chan_sip.c: Fix auth on BYE. (Different patch than for 1.2) * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE * channels/chan_sip.c: Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson) * channels/chan_sip.c: Issue #9439 - properly handle username parameters in SIP uri. * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!) ........ * /, channels/chan_sip.c: Merged following patch with a lot of changes for 1.4 ------ Merged revisions 64514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines Issue #9726 - rlister - Better logging for ACL denials While at it, also added better logging and handling of peers that are not supposed to register. My patch, stole the issue report from Russell. My apologies, Russell :-) ........ 2007-05-16 08:44 +0000 [r64515] Christian Richter * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode. ........ 2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant * res/res_features.c: Properly fix a problem that occurs when you set PARKINGEXTEN to an exten where a call is already parked. (issue #9723, patch by me) * res/res_features.c: When someone requests a specific parking space using the PARKINGEXTEN variable, ensure that no other caller is already there. (issue #9723, reported by mdu113, patch by me) 2007-05-14 19:26 +0000 [r64324] Olle Johansson * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit more 2007-05-14 19:13 +0000 [r64306] Russell Bryant * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication will trigger an error and cause sounds to stop, which in this case, is ringing. 2007-05-14 18:52 +0000 [r64280] Olle Johansson * channels/chan_sip.c: Handle network errors, like host or network unreachable, in a better way. This means that calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us that there is an issue. Since this is an unconnected UDP socket, we will not get error messages directly in most cases, but maybe on the second and third try. This is already implemented in trunk. 2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp * codecs/codec_speex.c: Properly set datalen field when doing PLC in codec_speex. (issue #9722 reported by mihai) * /, main/devicestate.c: Merged revisions 64275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 lines Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies) ........ * main/channel.c: Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold. 2007-05-14 13:58 +0000 [r64193] Steve Murphy * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant. 2007-05-14 10:39 +0000 [r64157] Olle Johansson * main/channel.c: Add hangupcause when we lack codecs for transcoding 2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp * channels/chan_sip.c: This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts? * channels/chan_sip.c: Tweak hold flags some more. They can be of three states when active: active, inactive, one direction. * channels/chan_sip.c: Ensure the onhold flag is set no matter what when being put on hold. 2007-05-11 20:16 +0000 [r63982] Jason Parker * main/manager.c: Hide manager password from "manager show user foo". I realize that there are other ways to get this, but we really don't need to just show it in plain text so easily. Issue 9273, patch by junky 2007-05-11 16:35 +0000 [r63905] Tilghman Lesher * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions 63903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) | 2 lines Issue 9121 - fixups for safe_asterisk script ........ 2007-05-11 16:05 +0000 [r63886] Russell Bryant * main/manager.c: When MD5 authentication is not possible because there is no challenge present, either because the Challenge action was never issued, or some other reason, give a proper error message and return an error instead of claiming that the user wasn't found. (reported by jsmith on IRC) 2007-05-11 15:43 +0000 [r63872] Joshua Colp * res/res_features.c: Make the PARKINGEXTEN feature of parking actually work. (issue #9708 reported by mdu113) 2007-05-10 23:15 +0000 [r63830] Jason Parker * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines Fix an issue with trying to kill a thread before it gets created. Issue 9709, patch by nic_bellamy. ........ 2007-05-10 22:23 +0000 [r63804] Russell Bryant * main/manager.c: Strip terminal escape sequences from CLI command output that is going to be sent out over the manager interface. (issue #9659, reported by pari, fixed by me) 2007-05-10 20:48 +0000 [r63750] Doug Bailey * main/callerid.c: Add test for negative offsets in cid data to prevent infinite loops. 2007-05-10 20:46 +0000 [r63749] Olle Johansson * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines Do not allocate SIP pvt's for PEERs we can not reach. This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel. ........ 2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp * main/channel.c: Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. * channels/chan_sip.c: Do not prematurely go on hold if sendonly was not actually set. 2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines Make sure we only create a DSP if it's requested on SUB_REAL ........ 2007-05-09 16:55 +0000 [r63612] Russell Bryant * main/channel.c: Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the code in that if a channel does not have a send_digit_begin() callback, it only cares about DTMF END events. (pointed out by Michael Neuhauser on the asterisk-dev list) 2007-05-09 16:54 +0000 [r63611] Joshua Colp * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister) ........ 2007-05-09 16:43 +0000 [r63608] Russell Bryant * main/channel.c: Only call ast_senddigit_begin() in ast_senddigit() if the channel has a send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the wrong thing to do, because that flag indicates that a *bridged* channel only wants DTMF END events coming from this channel. 2007-05-09 14:50 +0000 [r63566] Tilghman Lesher * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) | 2 lines Replicate fix from 51158 (app_voicemail) to app_directory (Issue 9224) ........ 2007-05-09 13:24 +0000 [r63535] Russell Bryant * Makefile: I have seen multiple people post questions trying to figure out what the message "The configure script must be executed before running 'make'" means. So, add another like that says to specifically run ./configure. If this isn't obvious enough, then they should be using something like AsteriskNOW and not installing from source. 2007-05-09 13:17 +0000 [r63534] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 62945,63402,63519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch. ........ r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while. ........ r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults. ........ 2007-05-09 13:04 +0000 [r63532] Olle Johansson * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users) 2007-05-08 22:38 +0000 [r63478] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) | 2 lines Issue 9602 - segfault in app_macro ........ 2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant * res/res_features.c: I mixed up the use of the find_feature() function, so I renamed it find_dynamic_feature, and changed the code to use the correct lock when using it. * res/res_features.c: Use a read/write lock when accessing the built-in features. * contrib/scripts/realtime_pgsql.sql (added), contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to contrib/scripts to be with the rest of the sql examples. (issue #9676, suretec) 2007-05-08 06:22 +0000 [r63360] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) | 2 lines Issue 9527 - upon entering a folder, no message is selected (curmsg == -1), so deleting causes memory corruption (beyond bounds) ........ 2007-05-07 22:28 +0000 [r63329] Russell Bryant * configs/res_pgsql.conf.sample (added), configs/extconfig.conf.sample, contrib/realtime_pgsql.sql (added): Add a sample configuration file and example tables for use with res_config_pgsql. (issue #9676, suretec) 2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged revisions 63285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek) ........ * channels/chan_sip.c: Minor backport of revision 59083 in trunk. Don't queue an unhold frame up if the call was never on hold to begin with. 2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson * main/config.c: Don't remove configuration from memory just because one section failed. * /: Guess svnmerge doesn't handle files that move around. Blocking patch to ./config.c 2007-05-06 12:28 +0000 [r63152] Olle Johansson * main/file.c: Stop the video stream when you stop playback of all streams for a call 2007-05-04 20:03 +0000 [r63099] Jason Parker * res/res_jabber.c: Fix a crash when checking version attribute in an incoming XML caps element. Issue 9667, patch by phsultan. 2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni * configs/manager.conf.sample: explanation for httptimeout in manager.conf 2007-05-03 16:44 +0000 [r62989] Joshua Colp * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes) ........ 2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming * main/loader.c: improve loader a bit, by avoiding trying to initialize embedded modules twice and avoiding trying to load modules from disk when they have been loaded already during the 'preload' pass (reported by blitzrage on IRC, patch by me) 2007-05-03 15:23 +0000 [r62942] Russell Bryant * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending). This set of changes came from a debugging session I had with Dwayne Hubbard. When he called into his home FXO, ran the Echo application, and pressed a digit, the digit would be echoed back and would never end. This is fixed, along with a couple other little improvements. * When chan_zap is in the middle of playing a digit to a channel, it feeds back null frames, not voice frames. So, I have modified ast_read to check the timing on emulated DTMF when it receives null frames, in addition to where it was doing this on voice frames. * Make a tweak to setting the duration on emulated DTMF digits. If there was no duration specified, it set it to be the minimum, instead of the default. * Instead of timing the emulated digits off of the number of samples in audio frames that pass through, just use time values. Now there is no code in this section that assumes 8kHz audio. 2007-05-03 14:41 +0000 [r62913] Steve Murphy * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19 (added), pbx/ael/ael-test/ael-test18/extensions.ael, pbx/ael/ael-test/ael-test19/extensions.ael (added), pbx/ael/ael-test/ael-test19 (added), pbx/ael/ael-test/ref.ael-test20 (added), pbx/ael/ael-test/ael-test20/extensions.ael (added), pbx/ael/ael-test/ael-test20 (added): updated the ael regressions to match what's in trunk 2007-05-03 14:36 +0000 [r62912] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged revisions 61357,61770,62885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line some fixes for PMP Hold/Retrieve, it should work now, when briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident. ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad. ........ 2007-05-03 13:54 +0000 [r62883] Steve Murphy * pbx/ael/ael-test/ref.ael-test18 (added), pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ael-test18/extensions.ael (added), pbx/ael/ael-test/ael-test18 (added), pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7: These mods fix bug 9623, where an '@' in the eswitch contents causes a syntax error. I also updated the regressions. 2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 May 2007) | 2 lines doh... initializing the pointer variable will work just a bit better ........ * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 May 2007) | 7 lines increase reliability and efficiency of static Realtime config loading via ODBC: don't request fields we aren't going to use don't request sorting on fields that are pointless to sort on explicitly request the fields we want, because we can't expect the database to always return them in the order they were created (reported by blitzrage in person (!), patch by me) ........ * res/res_config_pgsql.c: improve static Realtime config loading from PostgreSQL: don't request sorting on fields that are pointless to sort on use ast_build_string() instead of snprintf() don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway (patch by me, inspired by blitzrage's bug report about res_config_odbc) 2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant * main/channel.c: Merge changes from team/russell/inband_dtmf ... Fix some issues related to generating inband DTMF. There are two changes here: 1) The list of DTMF tones in the senddigit_begin() function explicitly specified 100ms of the tone followed by 100ms of silence. This really broke things with the way that Asterisk now wants complete control over when the digit begins and ends. So, regardless of what Asterisk really wanted to do, this was going to play out the tone at the length it wanted to. This caused various problems like DTMF translation to inband to be extremely unreliable. The list of tones has been changed so that the correct DTMF tone is played indefinitely until Asterisk tells it to stop. 2) ast_write() had to be modified to let a DTMF_END frame get processed even when a generator is present. This is how the tone will finally get stopped. (issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for the testing and feedback!) * main/manager.c: Backport the change that only went in to trunk that fixes the command manager action over http. (reported internally by pari and bkruse) 2007-05-02 20:46 +0000 [r62738] Steve Murphy * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being in 'h' extension louses up the dst field ........ 2007-05-02 17:43 +0000 [r62692] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines Issue 9638 - if a text frame is sent with no terminating NULL through a bridged IAX connection, the remote end will receive garbage characters tacked onto the end. ........ 2007-05-02 17:10 +0000 [r62689] Steve Murphy * configs/extensions.conf.sample, main/channel.c, main/pbx.c, channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS. 2007-05-02 06:15 +0000 [r62624] Olle Johansson * channels/chan_sip.c: Don't unlock a channel that we already know does not exist (propably isue 8228) 2007-05-01 21:57 +0000 [r62548] Russell Bryant * /, res/res_features.c: Merged revisions 62547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines Remove an unnecessary check that makes it so if you hang up after doing an attended transfer before the target extension answers the channel, the transfer is not successful. (issue #9338, patch by svanlund) ........ 2007-05-01 21:34 +0000 [r62545] Tilghman Lesher * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user() (found by rayjay, different fixes by me) 2007-05-01 16:26 +0000 [r62497] Russell Bryant * /, configs/indications.conf.sample: Merged revisions 62496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines Add indications.conf information for the Philippines. (issue #9525, reported and patched by loloski) ........ 2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines This patch fixes an issue where depending on the cause code, when the network sends a PRI disconnect, the call may not be properly hung up. (issue #9588, reported and patched by softins) ........ * include/asterisk/http.h, main/http.c: When serving dynamic content, include a Cache-Control header to instruct the browsers to not store the resulting content. (issue #9621, reported by Pari, patch by me) 2007-04-30 14:52 +0000 [r62371] Jason Parker * configs/iax.conf.sample: Remove unused (and potentially confusing) jitterbuffer options from sample config. 2007-04-30 14:36 +0000 [r62369] Joshua Colp * main/asterisk.c, /: Merged revisions 62368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 lines Update copyright notice. It's now the year 2007! ........ 2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant * channels/chan_zap.c: Fix a bug that made the "language" setting in zapata.conf not functional. (issue #9626, reported and fixed by sergee) * apps/app_meetme.c: Note that the "talker optimization" option will be enabled by default in 1.6 2007-04-27 Russell Bryant * Asterisk 1.4.4 released. 2007-04-27 21:10 +0000 [r62218] Russell Bryant * channels/chan_agent.c: Fix a weird problem where when a caller talking to someone sitting behind an agent channel sent a digit, the digit would be played to the agent for forever. This is because chan_agent always returned -1 from its send_digit_begin and _end callbacks. This non-zero return value indicates to the Asterisk core that it would like an inband DTMF generator put on the channel. However, this is the wrong thing to do. It should *always* return 0, instead. When the digit begin and end functions are called on the proxied channel, the underlying channel will indicate whether inband DTMF is needed or not, and the generator will be put on that one, and not the Agent channel. (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me) 2007-04-27 16:17 +0000 [r62174] Jason Parker * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines This transcoder message needn't be a NOTICE. I've seen it cause confusion more than a few times. ........ 2007-04-27 16:14 +0000 [r62171] Russell Bryant * main/pbx.c: If no variables were passed into pbx_substitute_variables_helper_full(), then don't even bother creating a temporary bogus channel, since that is only for allowing certain functions to operate on the variables as if they were on a channel. Most importantly, this fixes a crash. (issue #9613, reported by callguy, fixed by me) 2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka - THANKS!!!! THis was a hard one to catch. ........ * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks! 2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave) ........ * main/channel.c: Missed an ast_app_group_discard during merge. Thanks blitzrage! * res/res_monitor.c: Don't always say that the channel is being paused if it is actually being unpaused in the Manager ack message. (reported by jsmith in #asterisk-bugs) * main/config.c, /: Merged revisions 61958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 lines Don't count failed include attempts against the configuration include level. (issue #9593 reported by mostyn) ........ 2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286) ........ 2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines If the callerid= option is specified, but empty, clear any previous data. ........ * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines Ensure that callerid settings are reset on a reload. ........ 2007-04-25 19:21 +0000 [r61805] Joshua Colp * main/cli.c, main/channel.c, include/asterisk/app.h, funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh) ........ 2007-04-25 16:22 +0000 [r61799] Russell Bryant * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines Fix a typo where cid_num got copied instead of cid_ani. (issue #9587, reported and patched by xrg) ........ 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.