diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-20 20:33:34 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-20 20:33:34 +0000 |
| commit | 1afc2ddd7d179fcd6f2092c12990195a7e4a2832 (patch) | |
| tree | 579c8331e85ae1bc0c1f8272fd905241c3bda2d7 | |
| parent | d72f4c1ab145c404c7447264a85793b46ee6babb (diff) | |
| parent | c77c32c29eea06d4f3c00fc1feda9bdefcbca71b (diff) | |
Creating tag for the release of asterisk-1.2.26.1-netsec
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.2.26.1-netsec@94261 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 6513 | ||||
| -rw-r--r-- | channels/chan_iax2.c | 4 |
4 files changed, 2 insertions, 6517 deletions
diff --git a/.lastclean b/.lastclean deleted file mode 100644 index ec635144f..000000000 --- a/.lastclean +++ /dev/null @@ -1 +0,0 @@ -9 diff --git a/.version b/.version deleted file mode 100644 index ce2a1dc67..000000000 --- a/.version +++ /dev/null @@ -1 +0,0 @@ -1.2.26.1-netsec diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index 314931620..000000000 --- a/ChangeLog +++ /dev/null @@ -1,6513 +0,0 @@ -2007-12-20 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.26.1-netsec released - -2007-12-20 17:29 +0000 [r94214] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a couple of places where it's possible - to dereference a NULL pointer. - -2007-12-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.26-netsec released - -2007-12-18 18:44 +0000 [r93667-93675] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Fixing AST-2007-027 - (Closes issue #11119) - -2007-11-29 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.25-netsec released - -2007-11-29 21:10 +0000 [r90170] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c: Properly escape src and dst fields (Fixes - AST-2007-026) - -2007-09-13 18:10 +0000 [r82334] Kevin P. Fleming <kpfleming@digium.com> - - * LICENSE: clarify the OpenSSL and OpenH323 license exceptions - -2007-08-07 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.24-netsec released - -2007-08-07 17:44 +0000 [r78370] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Revert patch committed for issue #9660. It - broke E&M trunks. (closes issue #10360) (closes issue #10364) - -2007-08-02 17:56 +0000 [r77942] Steve Murphy <murf@digium.com> - - * fskmodem.c: 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. - -2007-07-31 19:19 +0000 [r77842] Steve Murphy <murf@digium.com> - - * contrib/scripts/ast_grab_core: This probably isn't super-general, - but it's a first stab at using kill -11 to generate a core file - instead of gcore. - -2007-07-30 18:40 +0000 [r77782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_agi.c: Revert change in revision 71656, even though it - fixed a bug, because many people were depending upon the (broken) - behavior. - -2007-07-30 14:50 +0000 [r77767] Joshua Colp <jcolp@digium.com> - - * apps/app_macro.c: (closes issue #10334) Reported by: ramonpeek - Pass through the return value from macro_exec through the MacroIf - application. - -2007-07-25 00:07 +0000 [r76978] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c: this fixes bug 10293, where the error - message because defaultzone or loadzone was not defined was - confusing - -2007-07-24 22:11 +0000 [r76934] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/lock.h: Oops, res contains the error code, not - errno. I was wondering why a mutex was reporting "No such file or - directory"... - -2007-07-24 Jason Parker <jparker@digium.com> - - * Asterisk 1.2.23-netsec released - -2007-07-24 16:32 +0000 [r76802] Jason Parker <jparker@digium.com> - - * channels/chan_iax2.c: Don't create the Asterisk channel until we - are starting the PBX on it. (ASA-2007-018) - -2007-07-23 18:28 +0000 [r76560-76653] Joshua Colp <jcolp@digium.com> - - * channels/chan_agent.c: (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. - - * channels/chan_sip.c: (closes issue #10236) Reported by: homesick - Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) - Accept Remote Party ID on guest calls. - -2007-07-22 21:39 +0000 [r76409] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/app.h: We should not use C++ reserved words in - API headers (closes issue #10266) - -2007-07-21 02:01 +0000 [r76226] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: 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 17:16 +0000 [r76080] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: (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. - -2007-07-19 15:49 +0000 [r75757-75927] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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) - - * channels/chan_iax2.c: 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 20:31 +0000 [r75748] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Store prior to copy (closes issue #10193) - -2007-07-18 17:48 +0000 [r75657] Dwayne M. Hubbard <dhubbard@digium.com> - - * apps/app_queue.c: removed the word 'pissed' from ast_log(...) - function call for BE-90 - -2007-07-17 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.22 released - -2007-07-17 20:57 +0000 [r75440-75449] Russell Bryant <russell@digium.com> - - * channels/chan_skinny.c: Properly check for the length in the - skinny packet to prevent an invalid memcpy. (ASA-2007-016) - - * channels/iax2-parser.h, channels/chan_iax2.c, - channels/iax2-parser.c: 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: 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) - -2007-07-16 20:46 +0000 [r75251-75304] Kevin P. Fleming <kpfleming@digium.com> - - * dns.c: provide proper copyright/license attribution for this - structure that was copied from a BSD-licensed header file long, - long ago... - - * Makefile: install the LICENSE file along with the music files - - * sounds/fpm-world-mix.mp3 (removed), sounds/moh/fpm-calm-river.mp3 - (added), Makefile, sounds/moh (added), - sounds/moh/fpm-world-mix.mp3 (added), sounds/moh/LICENSE (added), - sounds/fpm-sunshine.mp3 (removed), sounds/moh/fpm-sunshine.mp3 - (added), sounds/fpm-calm-river.mp3 (removed): move FreePlayMusic - files into a subdirectory, and include a license statement for - them - -2007-07-13 20:35 +0000 [r75107] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Fix a couple potential minor memory leaks. - load_moh_classes() could return without destroying the loaded - configuration. - -2007-07-13 20:10 +0000 [r75066] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: 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 20:07 +0000 [r75052-75059] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: 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: (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 15:51 +0000 [r74814] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Only print out a warning for situations - where it is actually helpful. (issue #10187 reported by denke) - -2007-07-11 22:53 +0000 [r74766] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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 21:12 +0000 [r74719] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: 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 18:33 +0000 [r74656] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c: 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 17:15 +0000 [r74587] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, channels/Makefile: Use some Makefile magic - to determine if linux/compiler.h is present. (issue #10174 - reported by francesco_r) - -2007-07-10 19:57 +0000 [r74373-74427] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: 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. - - * channels/chan_agent.c: 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 - - * dns.c: 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 <russell@digium.com> - - * Asterisk 1.2.21.1 released - -2007-07-10 15:37 +0000 [r74316] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix a small typo in description in of - Voicemail() application. Issue 10170, patch by casper. - -2007-07-10 15:30 +0000 [r74313] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c: Only use ESCAPE when LIKE is used. (issue - #10075, this part reported by jmls on IRC, patch by me) - -2007-07-10 14:48 +0000 [r74264] Joshua Colp <jcolp@digium.com> - - * app.c: Ensure the group information category exists before trying - to do a string comparison with it. (issue #10171 reported by - mlegas) - -2007-07-09 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.21 released - -2007-07-09 21:00 +0000 [r74165] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: 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) - -2007-07-09 20:18 +0000 [r74158] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: 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. - -2007-07-06 23:01 +0000 [r73678-73768] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: 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) - - * res/res_config_odbc.c: (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: (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 15:26 +0000 [r73674] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Fixed a bug wherein agents get stuck busy. - (issue 9618, reported by jiddings, patched by moi) closes issue - #9618 - -2007-07-05 22:11 +0000 [r73547] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: we shouldn't allow G.723.1 endpoints to use - VAD, just like we don't support it for G.729 - -2007-07-05 19:15 +0000 [r73315-73466] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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) - - * apps/app_chanspy.c, channel.c: Tweak spy locking. (issue #9951 - reported by welles) - - * channels/chan_local.c: 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: 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 14:50 +0000 [r73207-73252] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: bchannel configurations like - echocancel and volume control, need to be setuped on inbound - calls too. - - * channels/chan_misdn.c: bad bug in overlapdial case, we called - start_pbx multiple times, because the state wasn't changed.. - -2007-07-03 12:34 +0000 [r73052] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c: 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 08:04 +0000 [r73004] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed issue, that misdn_l2l1_check could - only be called from mISDN Source channels.. #9449 - -2007-07-02 17:58 +0000 [r72924] Jason Parker <jparker@digium.com> - - * say.c: Fix an issue with playing "oclock" multiple times in - French with 24 hour time format. Issue 10101 - -2007-07-01 23:51 +0000 [r72805] Russell Bryant <russell@digium.com> - - * pbx/pbx_spool.c: 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. - -2007-06-29 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.20 released - -2007-06-29 16:30 +0000 [r72629] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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 13:08 +0000 [r72585] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: 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. - -2007-06-27 23:24 +0000 [r72378] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c: Update documentation to clarify variable - usage with MixMonitor. (issue #9494 reported by netoguy) - -2007-06-27 23:22 +0000 [r72333-72373] Brett Bryant <bbryant@digium.com> - - * asterisk.c: 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. - - * asterisk.c: Reverted changes for earlier revisions 72259 to - 72261. Issue #9654, #10010 - -2007-06-27 22:43 +0000 [r72327] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Fix issue where queue log events might be - missing. (issue #7765 reported by mtryfoss) - -2007-06-27 21:06 +0000 [r72267] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: 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 20:43 +0000 [r72259] Brett Bryant <bbryant@digium.com> - - * asterisk.c: Fixes 100% load when controlling terminal disappears. - Issue #9654, #10010 - -2007-06-27 20:23 +0000 [r72256] Joshua Colp <jcolp@digium.com> - - * channel.c: 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 18:40 +0000 [r72184] Jason Parker <jparker@digium.com> - - * 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 13:22 +0000 [r72040-72099] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - 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.h, channels/misdn/isdn_lib.c: simplified - channel finding and locking a lot. removed unnecessary #ifdefed - areas. - - * channels/misdn/isdn_lib.c: isdn_lib.c didn't compile - - * channels/misdn/isdn_lib.c: 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. - -2007-06-26 17:49 +0000 [r71847] Jason Parker <jparker@digium.com> - - * Makefile: Don't try to install an init script that doesn't exist. - Reported to me on #asterisk on Freenode IRC. - -2007-06-26 12:25 +0000 [r71656-71750] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Issue 10062 - Trying to move a message - without selecting one first results in memory corruption - - * res/res_agi.c: Issue 10035 - handle_exec returns a result - inconsistent with all of the other AGI commands - -2007-06-25 01:02 +0000 [r71414] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Ignore other URIs after the first in a 300 - Multiple Choice response. (issue #10041 reported by homesick) - -2007-06-24 20:04 +0000 [r71358] Russell Bryant <russell@digium.com> - - * asterisk.c: Revert the patch from issue 9654 due to an unexpected - side effect - -2007-06-24 17:32 +0000 [r71288] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * manager.c, db.c: Issue 10043 - There is a legitimate need to be - able to set variables to the empty string. - -2007-06-22 16:02 +0000 [r71124] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Send an unhold indication when going off - hold. (issue #10036 reported by speedy) - -2007-06-22 14:52 +0000 [r71065] Jason Parker <jparker@digium.com> - - * file.c, res/res_agi.c: Fix a few silly usages of ast_playstream() - - it only ever returns 0... Issue 10035 - -2007-06-22 14:39 +0000 [r71064] Brett Bryant <bbryant@digium.com> - - * asterisk.c: 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 - -2007-06-21 22:29 +0000 [r70948] Steve Murphy <murf@digium.com> - - * cdr.c: 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 21:37 +0000 [r70898] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Don't explode if the gain option is - specified without a value. (issue #9274 reported by mfarver) - -2007-06-21 19:13 +0000 [r70804] Steve Murphy <murf@digium.com> - - * cdr/cdr_custom.c: 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 13:11 +0000 [r70672] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: we activate the - bchannels in TE mode on incoming calls only when we want to - connect the call. - -2007-06-20 22:20 +0000 [r70551] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't overwrite the configured username - setting upon a REGISTER. (issue #8565 reported by jsmith) - -2007-06-20 19:25 +0000 [r70444] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c: Issue 9997 - Timelimit times out the wrong - channel - -2007-06-20 18:45 +0000 [r70396] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: 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 15:42 +0000 [r70311-70342] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: forgot one place .. - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_lib.c: fixed a bug that was introduced by - copy and paste in the last commit ..bchannels weren't cleaned - properly. - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_lib.c: 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. - -2007-06-19 18:07 +0000 [r70053] Steve Murphy <murf@digium.com> - - * channel.c: 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 17:00 +0000 [r69992] Joshua Colp <jcolp@digium.com> - - * rtp.c: Handle the CC field in the RTP header. (issue #9384 - reported by DoodleHu) - -2007-06-19 16:45 +0000 [r69990] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: 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 16:21 +0000 [r69894-69986] Joshua Colp <jcolp@digium.com> - - * channel.c: Update BRIDGEPEER variable if set to the new channel - name when a masquerade happens. (issue #9699 reported by dimas) - - * apps/app_meetme.c: Perform an extra hangup check just in case. - (issue #9589 reported by bcnit) - -2007-06-19 13:23 +0000 [r69887] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: when we send out a SETUP, but get no - response, we should cleanup everything after reception of a - hangup. - -2007-06-19 12:57 +0000 [r69765-69846] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: 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) - - * channels/chan_sip.c: 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 17:45 +0000 [r69743] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/safe_asterisk: Issue 9998 - Remove SIG prefix, - since it's not supported by ksh - -2007-06-15 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.19 released - -2007-06-14 23:21 +0000 [r69469] Jason Parker <jparker@digium.com> - - * config.c: 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) - -2007-06-14 20:56 +0000 [r69347] Russell Bryant <russell@digium.com> - - * channel.c: Backport rev 69010 from the 1.4 branch ... 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) - -2007-06-14 15:15 +0000 [r69258] Jason Parker <jparker@digium.com> - - * funcs/func_groupcount.c: 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 18:12 +0000 [r69127] Joshua Colp <jcolp@digium.com> - - * app.c: Return group counting to previous behavior where you could - only have one group per category. (issue #9711 reported by - irroot) - -2007-06-13 09:55 +0000 [r69053] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_msg_parser.c: restart indicator 0x80 is - correct, at least that's what libpri does. - -2007-06-12 14:18 +0000 [r68921] Joshua Colp <jcolp@digium.com> - - * rtp.c: Bring RTP back to Asterisk at the end of a native bridge - no matter what. - -2007-06-12 08:35 +0000 [r68732-68887] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: if the bridged partner is mISDN too we - should not send dtmf tones, they are transmitted inband always - - * channels/chan_misdn.c: if we have already some digits, we just - stop the tones. - - * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: 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 14:29 +0000 [r68682] Joshua Colp <jcolp@digium.com> - - * channel.c: Improve deadlock handling of the channel list. (issue - #8376 reported by one47) - -2007-06-11 09:18 +0000 [r68631] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, - channels/misdn/isdn_lib.c: 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. - -2007-06-08 22:22 +0000 [r68526] Russell Bryant <russell@digium.com> - - * apps/app_dictate.c: 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 00:15 +0000 [r68368-68397] Joshua Colp <jcolp@digium.com> - - * say.c: 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) - - * dnsmgr.c: 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 23:13 +0000 [r68351] Russell Bryant <russell@digium.com> - - * say.c: 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 20:02 +0000 [r68204] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Don't try to save voicemail greetings - unless the user presses '1' to accept/save. Issue 9904, patch by - me. - -2007-06-07 14:19 +0000 [r67938-68070] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Allow the 'g' option to work if used with the - 'S' option. (issue #9888 reported by gasparz) - - * channels/chan_sip.c: Only notify the devicestate system of a peer - state change when the peer is built from the config file. (issue - #9900 reported by arkadia) - -2007-06-06 16:40 +0000 [r67715] Russell Bryant <russell@digium.com> - - * channel.c: 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 13:28 +0000 [r67593-67649] Joshua Colp <jcolp@digium.com> - - * rtp.c: Reinvite the RTP back to the Asterisk machine when the - timeout happens. (issue #9888 reported by gasparz) - - * devicestate.c: 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 15:42 +0000 [r67306-67307] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/chan_misdn_config.h: briding is a bool, fixed copy - and paste issue. - - * channels/chan_misdn.c: 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 11:18 +0000 [r67239] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h: Backport of the overlap_dial - functionality from asterisk-1.4's chan_misdn. - -2007-06-05 10:05 +0000 [r67209] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h: added possibility to - deactivate bridging per port - -2007-06-04 23:41 +0000 [r67060-67161] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_math.c: According to MATH, 0+1181000386 = 1181000448. - Oops. - - * 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: Add revision Id tags (by - request of tzafrir) - -2007-05-31 16:12 +0000 [r66764] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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) - -2007-05-31 15:58 +0000 [r66744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_macro.c: 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. - -2007-05-29 21:49 +0000 [r66537] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: If the value of a variable passed to - FIELDQTY is blank, then FIELDQTY should return 0, not 1. - -2007-05-29 07:53 +0000 [r66349] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #9802 - Change inuse counter on CANCEL - -2007-05-25 13:46 +0000 [r66127] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * say.c: Issue 9791 - Fix pronunciation of seconds in Dutch - -2007-05-24 14:40 +0000 [r65837] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 09:19 +0000 [r65767] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: 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 20:46 +0000 [r65676-65682] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: ensure that variables are set on a newly - created channel before we start a PBX on it - - * channels/chan_iax2.c: 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 13:06 +0000 [r65389-65588] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Revert revision 62417 as someone reported - problems with it to Mark. This was related to issue #9588. - - * apps/app_queue.c: 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 07:46 +0000 [r65328] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: 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 - -2007-05-18 20:56 +0000 [r65172] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, include/asterisk/cdr.h, cdr.c: 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 18:10 +0000 [r65075-65122] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Not getting an ACK to a 200 OK in the - initial invite is critical to the call. - - * channels/chan_sip.c: 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. - -2007-05-18 11:23 +0000 [r64902-65007] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: fixed a - warning regarding Keypad encoding. encode the IE sending_complete - at the right position. - - * channels/chan_misdn.c: we *need* to send a PROCEEDING when - sending_complete is set, even if need_more_infos is requested. - -2007-05-17 21:14 +0000 [r64819] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: How is it that we never caught - that this is returning the opposite of our documentation, until - now? - -2007-05-17 16:52 +0000 [r64758] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: If we have a negative current message, we - shouldn't go back even further... Issue 9727. - -2007-05-16 10:55 +0000 [r64514-64603] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Fixing possible bug in auth of BYE - - * channels/chan_sip.c: Support SIP uri's starting with SIP: and - sip: (reported by Tony Mountfield on the mailing list. Thanks!) - - * channels/chan_sip.c: 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 08:23 +0000 [r64513] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: 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. - -2007-05-14 18:34 +0000 [r64275] Joshua Colp <jcolp@digium.com> - - * devicestate.c: 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) - -2007-05-11 16:31 +0000 [r63903] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * Makefile, contrib/scripts/safe_asterisk: Issue 9121 - fixups for - safe_asterisk script - -2007-05-10 23:14 +0000 [r63828] Jason Parker <jparker@digium.com> - - * channels/chan_iax2.c: Fix an issue with trying to kill a thread - before it gets created. Issue 9709, patch by nic_bellamy. - -2007-05-10 20:38 +0000 [r63748] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: 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 17:20 +0000 [r63653] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we only create a DSP if it's - requested on SUB_REAL - -2007-05-09 16:51 +0000 [r63610] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 14:48 +0000 [r63565] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Replicate fix from 51158 (app_voicemail) to - app_directory (Issue 9224) - -2007-05-09 11:26 +0000 [r63519] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: release_chan frees ch, so we should never - touch ch after release_chan, this may cause segfaults. - -2007-05-08 22:19 +0000 [r63477] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_macro.c: Issue 9602 - segfault in app_macro - -2007-05-08 15:07 +0000 [r63402] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: 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. - -2007-05-08 06:20 +0000 [r63359] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Issue 9527 - upon entering a folder, no - message is selected (curmsg == -1), so deleting causes memory - corruption (beyond bounds) - -2007-05-07 21:39 +0000 [r63285] Joshua Colp <jcolp@digium.com> - - * include/asterisk/app.h, app.c, channel.c: Properly handle what - happens during a masquerade in relation to group counting. (issue - #9657 reported by ramonpeek) - -2007-05-07 14:06 +0000 [r63195] Olle Johansson <oej@edvina.net> - - * config.c: Releasing the whole in-memory configuration while - you're adding to it is not a good thing. Please review this - change. Caused by additions (+) to non-existing contexts. - -2007-05-03 16:42 +0000 [r62987] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 15:39 +0000 [r62885-62945] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: 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. - - * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: 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 00:23 +0000 [r62796-62841] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c: doh... initializing the pointer variable - will work just a bit better - - * res/res_config_odbc.c: 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) - -2007-05-02 20:10 +0000 [r62737] Steve Murphy <murf@digium.com> - - * cdr.c, pbx.c: Some tweaks to satisfy CDR bug 8796, where being in - 'h' extension louses up the dst field - -2007-05-02 17:38 +0000 [r62691] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_iax2.c: 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. - -2007-05-01 21:55 +0000 [r62496-62547] Russell Bryant <russell@digium.com> - - * res/res_features.c: 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) - - * configs/indications.conf.sample: Add indications.conf information - for the Philippines. (issue #9525, reported and patched by - loloski) - -2007-04-30 15:57 +0000 [r62417] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: 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) - -2007-04-30 14:34 +0000 [r62368] Joshua Colp <jcolp@digium.com> - - * asterisk.c: Update copyright notice. It's now the year 2007! - -2007-04-27 16:16 +0000 [r62173] Jason Parker <jparker@digium.com> - - * codecs/codec_zap.c: This transcoder message needn't be a NOTICE. - I've seen it cause confusion more than a few times. - -2007-04-27 13:57 +0000 [r62126] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: 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. - -2007-04-26 16:30 +0000 [r61958-62037] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: 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) - - * config.c: Don't count failed include attempts against the - configuration include level. (issue #9593 reported by mostyn) - -2007-04-25 22:24 +0000 [r61913] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: handle a very bizarre race condition with - channels being redirected before a simple switch can be started - on them (issue #9286) - -2007-04-25 21:55 +0000 [r61862-61866] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: If the callerid= option is specified, but - empty, clear any previous data. - - * channels/chan_iax2.c: Ensure that callerid settings are reset on - a reload. - -2007-04-25 18:52 +0000 [r61804] Joshua Colp <jcolp@digium.com> - - * include/asterisk/app.h, funcs/func_groupcount.c, app.c, - apps/app_groupcount.c, channel.c: 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 16:20 +0000 [r61798] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix a typo where cid_num got copied instead - of cid_ani. (issue #9587, reported and patched by xrg) - -2007-04-24 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.18 released - -2007-04-24 21:33 +0000 [r61786] Russell Bryant <russell@digium.com> - - * manager.c: 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) - -2007-04-24 18:20 +0000 [r61776-61777] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_phone.c, channels/chan_zap.c, - channels/chan_modem.c: removed #if 0 block from chan_phone, - chan_zap, and chan_modem restart_monitor() - - * channels/chan_modem.c: removed pthread_join in restart_monitor() - to make it like chan_zap and other channel drivers - -2007-04-24 16:05 +0000 [r61771] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Allow RFC2833 to be sent in the response SDP - when an INVITE comes in without SDP. (issue #9546 reported by - mcrawford) - -2007-04-24 13:50 +0000 [r61770] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, - channels/misdn/isdn_lib.c: 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. - -2007-04-20 21:14 +0000 [r61692-61704] Jason Parker <jparker@digium.com> - - * loader.c: 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. - - * apps/app_queue.c: 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 - -2007-04-19 07:56 +0000 [r61685] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Send NOTIFY to Contact: in SUBSCRIBE - as - reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp. - -2007-04-19 02:30 +0000 [r61680] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_language.c, funcs/func_moh.c, funcs/func_groupcount.c, - funcs/func_timeout.c, funcs/func_cdr.c, funcs/func_callerid.c: - 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. - -2007-04-16 14:08 +0000 [r61663] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't stop RTP on errors on INFO messages. - Disclaimer: This patch was needed for Edvina AstHoloApp and was - meant to be included in 1.2, but never made it in time so I felt - I could add it now. No, just joking, patching error found while - testing T.140 with Omnitor earlier this spring. - -2007-04-13 19:15 +0000 [r61655] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: 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) - -2007-04-11 16:01 +0000 [r61376-61476] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: 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) - - * channels/chan_sip.c: 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) - - * channels/chan_sip.c: 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 12:05 +0000 [r61357] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: some fixes for - PMP Hold/Retrieve, it should work now, when briding=no - -2007-04-10 12:31 +0000 [r61170] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c: msns config parameter defaults to '*' - -2007-04-09 19:38 +0000 [r61038] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Don't send ActionID before Response: - header. - Don't use a blank in an AMI header - -2007-04-09 17:22 +0000 [r60935] Jason Parker <jparker@digium.com> - - * apps/app_directory.c: 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 02:49 +0000 [r60846-60849] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk.h, asterisk.c: 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: Bug 9505 - If the return value for - local_queue_frame is set, then p->lock is no longer valid. - -2007-04-09 00:59 +0000 [r60797] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: 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) - -2007-04-08 14:00 +0000 [r60660-60711] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_macro.c: Gosub called within a Macro resets the - arguments improperly and causes general weirdness. (Issue 8329) - - * file.c: Bug 9486 - memory leak when opening a filestream - -2007-04-06 17:03 +0000 [r60456] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: There should only be one code path for doing - DTMF conditionals on channels. This fixes it. - -2007-04-06 14:41 +0000 [r60398] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: remove undocumented 'cardsmode' parameter and - stop searching for transcoders during reload() - -2007-04-05 16:09 +0000 [r60267] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: 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 12:52 +0000 [r60213] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 17:38 +0000 [r59886-60134] Russell Bryant <russell@digium.com> - - * manager.c: 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: 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) - - * channels/chan_sip.c: 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) - - * mkpkgconfig: Use the more generic check for "sed -r" support that - was already present in 1.4. (related to issue #9399) - - * mkpkgconfig: On Darwin, the -r argument to sed is not valid. It - has to be -E. (issue #9399, reported by jcovert) - - * channels/chan_sip.c: 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: Make chan_sip report when it encounters an - unknown option. (issue #9440, reported by nightcrawler) - - * app.c: 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) - -2007-04-03 10:40 +0000 [r59788-59803] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c: ptp is the 5th bit, not the 4th. - - * channels/misdn_config.c, channels/misdn/chan_misdn_config.h: Use - the new sysfs way of mISDN 1.2 to check if a port is NT or not. - -2007-04-02 18:55 +0000 [r59723] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Increase the maximum size for a string of - mailboxes to 1024. (issue #9270 reported by rtucker) - -2007-04-02 12:08 +0000 [r59623-59639] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h: 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 - - * channels/chan_misdn.c: don't be verbose if no need - - * channels/misdn/isdn_lib.c: we can now make 30 channels on a PRI - (before we forgot chan 31..) - -2007-04-01 22:35 +0000 [r59608] Russell Bryant <russell@digium.com> - - * netsock.c: 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.) - -2007-03-29 17:33 +0000 [r59360] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: 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 17:14 +0000 [r59355-59357] Russell Bryant <russell@digium.com> - - * rtp.c: 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) - - * channels/chan_iax2.c: 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. - -2007-03-29 16:04 +0000 [r59299-59301] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_odbc.c: 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.) - - * res/res_odbc.c: Change ENV section to use setenv, instead of - putenv (Alexandru Pirvulescu <sigxcpu@gmail.com>, reported via - -dev list) - -2007-03-27 23:36 +0000 [r59280-59283] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Oops - - * apps/app_voicemail.c: Fix a few remaining bad mmap(2) return - values - -2007-03-27 23:19 +0000 [r59258-59277] Russell Bryant <russell@digium.com> - - * apps/app_directory.c: Fix the check of the return value from - mmap(). Thanks to Corydon for catching this one. - - * channels/chan_iax2.c: 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) - -2007-03-27 13:56 +0000 [r59252] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed #9355 - -2007-03-26 10:21 +0000 [r59199] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, channels/misdn/Makefile, - channels/misdn/chan_misdn_config.h, channels/Makefile, - channels/misdn/isdn_lib.c: mISDN >= 1.2 provides a dsp pipeline - for i.e. echo cancellation modules, make chan_misdn use it. - -2007-03-24 01:35 +0000 [r59194] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only try to handle a response if it has a - response code. (ASA-2007-011) - -2007-03-23 16:07 +0000 [r59186-59187] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: Ugh. that was dumb. Fixed an error. - - * apps/app_macro.c: Added a few words in the Macro doc strings - about the behavior of macros with hangups (et al.), as per 9337 - -2007-03-21 18:03 +0000 [r59086] Joshua Colp <jcolp@digium.com> - - * res/res_monitor.c: Indicate the filename changed when it is - changed. (issue #9311 reported by jsmith) - -2007-03-20 09:23 +0000 [r59062-59063] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: modified a loglevel - - * channels/chan_misdn.c: avoid sending a disconnect when we already - received one. - -2007-03-19 Jason Parker <jparker@digium.com> - - * Asterisk 1.2.17 released - -2007-03-14 16:38 +0000 [r58896] Russell Bryant <russell@digium.com> - - * SECURITY: 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. - -2007-03-13 12:58 +0000 [r58849-58850] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed the crypt_keys stuff - - * channels/misdn_config.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, - channels/misdn/isdn_lib.c: added method standard_dec for dialing - out on groups, to avoid conflicts, which caused issues with some - ISDN providers - -2007-03-13 11:45 +0000 [r58847] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #9229 - No port in request URI on - register to non default SIP ports (neelakantan) - -2007-03-12 16:49 +0000 [r58832] Joshua Colp <jcolp@digium.com> - - * include/asterisk/lock.h: We can't use the assembler version of - fetchadd_int under Intel Macs. (issue #9254 reported by darrell - budic) - -2007-03-09 20:46 +0000 [r58579] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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. - -2007-03-09 14:43 +0000 [r58558] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: we can free channel 31 as well, since - we can occupy it - -2007-03-08 16:04 +0000 [r58388] Joshua Colp <jcolp@digium.com> - - * dsp.c: Only print out debug message if the definition that makes - the variables shows up was actually defined. (issue #9233 - reported by serginuez) - -2007-03-07 18:17 +0000 [r58164-58242] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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. - - * manager.c: Fix a misplaced block of code in the 1.2 version of - the patch to fix issue #8977 - - * manager.c: 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 22:52 +0000 [r58115] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: 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 20:33 +0000 [r58052] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change error message to proper message - -2007-03-06 15:17 +0000 [r58008] BJ Weschke <bweschke@btwtech.com> - - * channels/h323/Makefile: Cleanup the Makefile so that we only - attempt to include a file when we're building and not 'clean'ing - so 'make clean' completes successfully. chan_h323 maintainer: - please check to make sure I haven't broken your build target. - From: jsmith in #asterisk-dev - -2007-03-05 23:18 +0000 [r57962] Christian Richter <christian.richter@beronet.com> - - * channels/Makefile: subdirs like misdn and h323 should be cleaned - by the clean target from channels/Makefile as well - -2007-03-05 17:49 +0000 [r57869] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Make create_dirpath use our standard for - return values. -1 is failure, 0 is success. (issue #9205 reported - by ballares) - -2007-03-05 14:53 +0000 [r57825] Steve Murphy <murf@digium.com> - - * pbx.c: Fixed a typo introduced via 9156 - -2007-03-04 10:39 +0000 [r57753] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed another - place where the out_cause was hardcoded to 16 - -2007-03-03 06:36 +0000 [r57648] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Memory leak of a list, if call recording - was abandoned - -2007-03-02 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.16 released - -2007-03-02 18:32 +0000 [r57523] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: fixed typo - -2007-03-02 17:02 +0000 [r57475] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 16:39 +0000 [r57458] Steve Murphy <murf@digium.com> - - * pbx.c: further refinement in wording of goto documentation, as - per 9156, goto not proceeding to next instruction - -2007-03-01 22:19 +0000 [r57317] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: 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) - -2007-02-28 19:12 +0000 [r57118] Steve Murphy <murf@digium.com> - - * pbx.c: 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 18:55 +0000 [r57092] Joshua Colp <jcolp@digium.com> - - * channels/chan_agent.c: Fix a few more issues with the agent - logoff CLI command. (issue #9123 reported by arbrandes) - -2007-02-28 16:09 +0000 [r57034] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed - bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302, - #303, #304 - -2007-02-26 20:05 +0000 [r56850] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_config.c: Obey the clearglobalvars option in extensions - reload (or dialplan reload depending on your version). (issue - #9146 reported by ramonpeek) - -2007-02-26 00:34 +0000 [r56729] Russell Bryant <russell@digium.com> - - * utils.c: 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 14:38 +0000 [r56684] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channel.c: 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. - -2007-02-23 23:20 +0000 [r56406-56504] Russell Bryant <russell@digium.com> - - * asterisk.c: 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. - - * channels/chan_iax2.c: 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). - -2007-02-22 23:19 +0000 [r56230-56279] Joshua Colp <jcolp@digium.com> - - * channels/chan_agent.c: Always defer Agent logoff if any channels - are up until they hang up. (issue #9123 reported by arbrandes) - - * channels/chan_sip.c, channel.c: 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 00:53 +0000 [r56010] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: If we receive a frame that is not in any of - the negotiated formats, then drop it. (potentially issue #8781 - and SPD-12) - -2007-02-21 20:32 +0000 [r55956] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: 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 14:03 +0000 [r55868] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_svn_branch_name: use new tag version script - -2007-02-21 00:19 +0000 [r55750] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c, utils.c, include/asterisk/lock.h: 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). - -2007-02-20 22:39 +0000 [r55669] Joshua Colp <jcolp@digium.com> - - * channels/chan_agent.c: 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 19:49 +0000 [r55588] Russell Bryant <russell@digium.com> - - * apps/app_page.c: Convert a tab to spaces so that the - documentation is printed out properly aligned. - -2007-02-19 18:09 +0000 [r55434] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: forcename and forcegreetings options should - check to see if the recording already exists - -2007-02-18 12:32 +0000 [r55249-55277] Olle Johansson <oej@edvina.net> - - * apps/app_record.c: Documentation update (#9053, jsmith) - - * channels/chan_sip.c: Issue #9020 - SIP message retransmission - time too short. Backporting fix implemented in 1.4, where we have - a minimum level for the T1 timer. - -2007-02-17 03:53 +0000 [r55073-55153] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Answer the channel before recording privacy - information. (issue #8926 reported by lmamane) - - * channels/chan_sip.c: 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) - -2007-02-16 22:48 +0000 [r55005] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c, configs/meetme.conf.sample: 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 22:13 +0000 [r54999] Joshua Colp <jcolp@digium.com> - - * channels/chan_agent.c: 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 20:56 +0000 [r54955] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: 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 11:38 +0000 [r54771] Olle Johansson <oej@edvina.net> - - * res/res_agi.c: Issue #9069 - If we open with TH we should not - close with /TD. (seanbright) - -2007-02-15 16:14 +0000 [r54622] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Use a separate variable to indicate execution - should continue instead of the return value. (issue #8842 - reported by pluto70) - -2007-02-14 18:40 +0000 [r54438] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - -2007-02-14 16:25 +0000 [r54373] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: When handling glare on a PRI, move the - requested channel rather than hang up the old one. Fix for 8957 - and 9011. - -2007-02-13 18:35 +0000 [r54179] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Make sure that outbound calls are applied to - the peer. This fixes some issues with "hints not working", but - only in 1.2. - -2007-02-08 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.15 released - -2007-02-08 22:17 +0000 [r53658] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: ensure channelcount is cleared before we - enumerate transcoders, so 'reload' doesn't double the channel - count - -2007-02-08 13:36 +0000 [r53529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Issue 9003 - If fullname is empty, quote() - passes back "\"" - -2007-02-07 15:38 +0000 [r53357] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Fix a few potential memory leaks with - realtime users and peers. (issue #8999 reported by bsmithurst) - -2007-02-07 15:30 +0000 [r53354] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_macro.c: Issue 7440 - Macro called from Macro from the h - extension exits prematurely - -2007-02-06 06:58 +0000 [r53245] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * manager.c: Issue 8987 - Status could return two responses - (mnicholson) - -2007-02-06 00:08 +0000 [r53224] Jason Parker <jparker@digium.com> - - * configs/dnsmgr.conf.sample: Add a proper newline at the end of - this sample config file. - -2007-02-03 20:39 +0000 [r53133-53134] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c, utils.c, include/asterisk/lock.h: Revert some - changes that accidentally got committed as a part of another fix. - - * apps/app_dial.c, apps/app_meetme.c, utils.c, - include/asterisk/lock.h: 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) - -2007-02-02 16:58 +0000 [r53117] Joshua Colp <jcolp@digium.com> - - * config.c: 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) - -2007-02-01 23:14 +0000 [r53107] Jason Parker <jparker@digium.com> - - * apps/app_chanspy.c: Fix a small typo. Synopsis lines shouldn't - have a newline - -2007-02-01 22:21 +0000 [r53095-53103] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Copy noncodeccapability over to the joint - variable so that telephone-event will get transmitted in the sent - INVITE. - - * channels/chan_sip.c: Don't negotiate RFC2833 when not configured - to do so. (issue #8799 reported by mdu113) - -2007-02-01 21:12 +0000 [r53090] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Make sure we release call from call - counter before we destroy call (maybe #7744 and more) - - Backported by accident from 1.4 - -2007-02-01 21:03 +0000 [r53084] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Return previous behavior of having MOH - pick up where it was left off. (issue #8672 reported by - sinistermidget) - -2007-02-01 20:07 +0000 [r53069-53074] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Bug 8965 - - * funcs/func_strings.c, pbx.c: No wonder FIELDQTY doesn't work with - functions... the documentation in pbx.c was wrong - -2007-01-31 21:25 +0000 [r53045] Russell Bryant <russell@digium.com> - - * channels/chan_mgcp.c, manager.c, channels/chan_zap.c, - pbx/pbx_spool.c, apps/app_meetme.c, apps/app_page.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_queue.c, - channels/chan_iax2.c, apps/app_rpt.c, cdr.c, pbx.c: Fix a bunch - of places where pthread_attr_init() was called, but - pthread_attr_destroy() was not. - -2007-01-31 18:58 +0000 [r53044] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: update to match modified transcoder API - -2007-01-31 17:41 +0000 [r53039] Russell Bryant <russell@digium.com> - - * rtp.c: Use the proper format string to print unsigned values in - the rtp debug output. (issue #8954, wmis) - -2007-01-31 17:28 +0000 [r53034] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/Makefile: allow codec_zap to build again, now that - transcoder support is in zaptel 1.2 - -2007-01-30 19:41 +0000 [r52857-52954] Russell Bryant <russell@digium.com> - - * channel.c: 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) - - * asterisk.c: 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. - - * codecs/Makefile: 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 14:38 +0000 [r52843] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some - possible segfaults. also fixed an very important bug which occurs - on high load (when calls are very fast generated) - -2007-01-30 00:15 +0000 [r52762] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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 23:39 +0000 [r52716] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c: 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 16:48 +0000 [r52503] Jason Parker <jparker@digium.com> - - * codecs/codec_zap.c: Use the correct zaptel header file location. - Currently, this will not build - transcoder support will be added - to zaptel later today. - -2007-01-27 02:09 +0000 [r52360-52415] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Make COMPLETECALLER and COMPLETEAGENT output to - queue_log follow documentation. (issue #7677 reported by amilcar) - - * channels/chan_iax2.c: Make the last context entry read in the - dominant one. (issue #8918 reported by pj) - -2007-01-25 19:15 +0000 [r52162-52264] Joshua Colp <jcolp@digium.com> - - * jitterbuf.c: Allow dequeueing of frames with negative timestamp - by moving jitterbuffer frames check to jb_next. (issue #8546 - reported by harmen) - - * apps/app_mixmonitor.c: Add another note about audio files being - played back to each bridged party. (issue #8718 reported by ppyy) - -2007-01-25 00:39 +0000 [r52137] Russell Bryant <russell@digium.com> - - * apps/app_groupcount.c: Fix a seg fault when running this - application with no arguments from AGI. (issue #8905, junky) - -2007-01-24 17:43 +0000 [r52002] Steve Murphy <murf@digium.com> - - * utils/check_expr.c, utils/Makefile: 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 10:48 +0000 [r51966] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed the busy problem (dialstatus was not - busy when we called a busy extension) - -2007-01-24 00:57 +0000 [r51828-51843] Russell Bryant <russell@digium.com> - - * channel.c: 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: 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 01:41 +0000 [r51512] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: 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 19:39 +0000 [r51410] Russell Bryant <russell@digium.com> - - * codecs/Makefile, codecs/codec_zap.c (added): Merge codec_zap - support for the transcoder card. This is a standalone codec - module so it will not affect anything else. - -2007-01-22 19:08 +0000 [r51359-51406] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c: 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) - - * channels/chan_h323.c: 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) - -2007-01-19 16:44 +0000 [r51300] Russell Bryant <russell@digium.com> - - * asterisk.c: 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) - -2007-01-18 23:47 +0000 [r51271] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c: 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 23:35 +0000 [r51269] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: support echo cancellers that can handle 64ms - or 128ms of echo cancellation - -2007-01-18 21:11 +0000 [r51235-51255] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * stdtime/localtime.c: If a timezone is not specified, assume - localtime (instead of gmtime) (Issue #7748) - - * contrib/scripts/vmdb.sql: Document all the fields, including the - indication that "uniqueid" should not be renamed. - -2007-01-17 21:17 +0000 [r51197] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: 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 20:52 +0000 [r51158-51194] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * utils.c: 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). - - * doc/voicemail_odbc_postgresql.txt (added): Add documentation - walkthrough on getting Postgres to work with voicemail (from - Issue 8513) - - * apps/app_voicemail.c: Postgres driver doesn't like a NULL pointer - when retrieving the length (Bug 8513) - -2007-01-16 17:36 +0000 [r51085-51145] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: 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) - - * channels/chan_zap.c: 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) - -2007-01-15 23:09 +0000 [r50987] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_groupcount.c: Check return value before dereferencing - (Bug 8822) - -2007-01-15 20:44 +0000 [r50946] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: Solves issue with forwarding voicemails - from folders other than inbox. patch by anthonyl. - -2007-01-14 05:01 +0000 [r50781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * db1-ast/recno/rec_utils.c, db1-ast/hash/hash_bigkey.c, - db1-ast/hash/hsearch.c, db1-ast/recno/rec_open.c, - db1-ast/recno/rec_delete.c, db1-ast/btree/bt_page.c, - db1-ast/hash/hash_buf.c, db1-ast/hash/hash_page.c, - db1-ast/recno/rec_close.c, db1-ast/recno/rec_search.c, - db1-ast/btree/bt_get.c, db1-ast/hash/hash.c, - db1-ast/recno/rec_put.c, db1-ast/include/ndbm.h, db1-ast/db/db.c, - db1-ast/btree/bt_debug.c, db1-ast/mpool/mpool.c, - db1-ast/btree/bt_seq.c, db1-ast/recno/rec_get.c, - db1-ast/btree/bt_split.c, db1-ast/hash/hash_func.c, - db1-ast/btree/bt_utils.c, db1-ast/btree/bt_open.c, - db1-ast/recno/rec_seq.c, db1-ast/btree/bt_delete.c, - db1-ast/btree/bt_overflow.c, db1-ast/hash/hash_log2.c, - db1-ast/btree/bt_search.c, db1-ast/btree/bt_conv.c, - db1-ast/btree/bt_close.c, db1-ast/btree/bt_put.c: Bug 8814 - db - should look for its header using a relative path, instead of the - system path (Fixes FreeWRT) - -2007-01-12 14:34 +0000 [r50561] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: minor documentation clarification - -2007-01-11 18:11 +0000 [r50517] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8793 bad response for Unsupported - Extension (different fix). - -2007-01-11 14:45 +0000 [r50495-50506] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: when we get L2 UP, the L1 is UP - definitely too, so we set the L1 state up as well. - - * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c, - channels/misdn/isdn_lib.c: * 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) - -2007-01-10 09:51 +0000 [r50335] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/fac.c, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: more - fixes regarding warnings for gcc-4 and first additions for the - restart Information element, in the first step we initiate a - restart with a CLI command - -2007-01-10 04:51 +0000 [r50295] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: 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) - -2007-01-10 02:16 +0000 [r50227] Russell Bryant <russell@digium.com> - - * Makefile: 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 13:30 +0000 [r50150] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: 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) - -2007-01-08 08:37 +0000 [r49922] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/misdn/ie.c, - channels/misdn/isdn_lib.c: make gcc 4 happy, remove some warnings - -2007-01-08 05:10 +0000 [r49889] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Ensure we use the default refresh value of - 60 if the remote server does not send one. (issue #8746 reported - by maethor) - -2007-01-07 21:43 +0000 [r49833] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dictate.c: If openstream fails, then we crash (Issue - 8564) - -2007-01-05 16:56 +0000 [r49635] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_iax2.c: ensure that threads which are supposed to - be detached (because we aren't going to wait on them) are created - properly - -2007-01-04 17:45 +0000 [r49354-49447] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: converted a lot of 256 to PATH_MAX and some - white space fixes. - - * apps/app_voicemail.c: good catch russell sorry i missed that. fix - magic number with proper sizeof - - * apps/app_voicemail.c: 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 08:24 +0000 [r49135-49303] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: * 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 - - * channels/misdn_config.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, configs/misdn.conf.sample, - channels/misdn/isdn_lib.c: 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. - -2006-12-29 00:32 +0000 [r49045] Kevin P. Fleming <kpfleming@digium.com> - - * BUGS: location of the bug posting guidelines has changed - -2006-12-27 15:43 +0000 [r48974] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8596: Set CAN_BYE flag for 100 trying - too - -2006-12-25 05:19 +0000 [r48939-48955] Russell Bryant <russell@digium.com> - - * funcs/func_math.c: 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_iax2.c: 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: Remove a couple of misplaced dots in log - messages. This was reported by Andrea Spadaccini on the - asterisk-dev mailing list. - -2006-12-21 20:25 +0000 [r48782] Joshua Colp <jcolp@digium.com> - - * redhat/asterisk.spec: Add new silence sound files to the spec for - Redhat. (issue #8652 reported by alvaro_palma_aste) - -2006-12-19 21:10 +0000 [r48584] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Free localuser structure when we fail to dial - (issue #8612 reported by rizzo) - -2006-12-19 13:08 +0000 [r48576] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: 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 - -2006-12-18 10:19 +0000 [r48552] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: 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. - -2006-12-15 10:51 +0000 [r48484] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8592 - handle 504 as 503 - congestion - -2006-12-14 13:03 +0000 [r48467] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: 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. - -2006-12-14 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.14 released - -2006-12-13 04:23 +0000 [r48434] Steve Murphy <murf@digium.com> - - * channel.c: This small patch fixes bug 8541, where the L option to - the Dial app wasn't working right. A similar bug (8386) was filed - and fixed earlier, but an intervening bug fix to a DTMF problem - broke the L() code in a different way. Hopefully, everything is - happy now. - -2006-12-12 05:11 +0000 [r48403] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/silence (added), sounds/silence/1.gsm (added), - sounds/silence/10.gsm (added), sounds/silence/2.gsm (added), - sounds/silence/3.gsm (added), sounds/silence/4.gsm (added), - sounds/silence/5.gsm (added), sounds/silence/6.gsm (added), - sounds/silence/7.gsm (added), sounds/silence/8.gsm (added), - sounds/silence/9.gsm (added): add silence files - -2006-12-11 23:00 +0000 [r48394-48398] Matt O'Gorman <mogorman@digium.com> - - * Makefile, apps/app_externalivr.c, sounds.txt: 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 00:33 +0000 [r48374] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * 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: 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 02:14 +0000 [r48371] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c: This version applies the patch suggested by - stevens in bug 7836 (make inbound channel RINGING state - consistent with other channels). - -2006-12-09 15:45 +0000 [r48361] Russell Bryant <russell@digium.com> - - * 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. - -2006-12-07 18:14 +0000 [r48356] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Ensure that the file position is not - incremented beyond the total number of files available for - playback. (issue #8539, ulogic) - -2006-12-06 16:05 +0000 [r48322] Russell Bryant <russell@digium.com> - - * configs/iax.conf.sample: Fix the name of the rtignoreregexpire - option in the sample configuration file. (issue #8526, arkadia) - -2006-12-06 15:48 +0000 [r48321] Christian Richter <christian.richter@beronet.com> - - * doc/README.misdn, channels/chan_misdn.c, - channels/misdn/isdn_msg_parser.c: 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. - -2006-12-06 15:42 +0000 [r48320] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8528 - make sure we don't delete the - dialog too quickly after receiving a 487. Move 487 handling into - handle_response_invite where it really belongs and don't add an - ALREADYGONE flag to the dialog. - -2006-12-06 14:35 +0000 [r48319] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: changed a few debugs to higher debug - levels - -2006-12-06 12:14 +0000 [r48272-48315] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't add Contact header on BYE, CANCEL, - MESSAGE requests (Bye, Cancel backported from 1.4, MESSAGE bug - reported to me by Gunnar at Omnitor) - - * channels/chan_sip.c: Only set the ALREADYGONE flag once in - handle_response() - -2006-12-05 01:26 +0000 [r48251] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: 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 21:20 +0000 [r48236-48246] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: 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. - - * apps/app_voicemail.c: 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 18:14 +0000 [r48233] Joshua Colp <jcolp@digium.com> - - * channel.c: 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. - -2006-12-01 23:30 +0000 [r48192] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c: 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 20:19 +0000 [r48183] Jason Parker <jparker@digium.com> - - * configs/extensions.conf.sample: Fix a small typo - issue 8848, - reported by pabelanger - -2006-11-30 20:47 +0000 [r48165] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8319 - noriyuki - nonce-count updated - *after* use - -2006-11-30 20:27 +0000 [r48142-48161] Joshua Colp <jcolp@digium.com> - - * channel.c: 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: Only print out debug message if bridged - channel is not NULL. (issue #8412 reported by jubilex) - - * res/res_features.c: 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) - - * cdr.c: 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) - - * res/res_features.c: 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: Document 'port' for SIP peers, came up - because of the current mailing list thread. (issue #8450 reported - by blitzrage) - -2006-11-30 09:05 +0000 [r48127] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Do proper test and don't leave dialogs - hanging... - -2006-11-29 16:47 +0000 [r48053-48106] Joshua Colp <jcolp@digium.com> - - * rtp.c: If the frame was duplicated before writing out then we - need to free it. (issue #8429 reported by edguy3) - - * channels/chan_phone.c: 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: Use the proper function to get the new - message count instead of always using the filesystem. (issue - #8421 reported by slimey) - -2006-11-27 17:15 +0000 [r48045] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_musiconhold.c: Random MOH wasn't really random (bug 8381) - -2006-11-27 15:30 +0000 [r48037] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_spool.c: Do not reference the freed outgoing structure in - the debug message. (issue #8425 reported by arkadia) - -2006-11-24 14:33 +0000 [r47987] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change some logging levels. Not having hints - is not an ERROR, but still should be reported. - -2006-11-23 16:10 +0000 [r47968] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/isdn_lib.c: fixed a litle bug regarding - HOLD/RETRIEVE. beatufied some logs, changed some loglevels. - changed the default value of block_on_alarm - -2006-11-23 10:54 +0000 [r47958] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Remove unused variable (rizzo) - -2006-11-22 02:19 +0000 [r47910] Steve Murphy <murf@digium.com> - - * channel.c: This is the fix for bug 8386, wherein the time-limit - args to dial didn't work correctly - -2006-11-20 19:59 +0000 [r47862] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Failing to trap -1 error from mmap causes - segfault (Issue 8385) - -2006-11-20 19:50 +0000 [r47855-47859] Joshua Colp <jcolp@digium.com> - - * frame.c: Don't forget to byte swap if we are exiting the smoother - feed early. (issue #8287 reported by arturs) - - * channels/chan_sip.c: Free history items at the end of use of the - temporary SIP pvt structure. (issue #8383 reported by benh) - -2006-11-20 10:17 +0000 [r47842] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Just to be safe, disable all the scheduled - items after deleting a scheduler entry (rizzo) - -2006-11-17 19:02 +0000 [r47802] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: backport proper channel_find_locked behavior from 1.4 - branch (noted by Steve Davies on asterisk-dev list) - -2006-11-16 23:16 +0000 [r47780] Jason Parker <jparker@digium.com> - - * apps/app_dial.c, apps/app_cut.c, apps/app_directory.c, - apps/app_db.c: Fix a couple of typos in applications.. Initially - spotted by mrobinson. - -2006-11-16 22:57 +0000 [r47776] Kevin P. Fleming <kpfleming@digium.com> - - * doc/README.cdr: update clearly wrong documentation regarding - cdr_custom - -2006-11-16 20:29 +0000 [r47750-47761] Joshua Colp <jcolp@digium.com> - - * cdr/Makefile: Look for the header file specifically in all cases, - not just the existence of the directory. (issue #8358 reported by - mrness) - - * channels/chan_local.c: 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) - -2006-11-16 16:44 +0000 [r47743] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't fixup if we haven't got PVT. - Suggestion from Martin Vit on -dev mailing list inspired by - file's commit to chan_local. "This shouldn't happen" ;-) - -2006-11-15 22:29 +0000 [r47711] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: 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 21:18 +0000 [r47705] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: CANCEL requests are never authenticated - (according to RFC 3261) - -2006-11-15 20:30 +0000 [r47666-47696] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: correct argument name typo that caused - global variable to be used instead of the one for the specified - voicemail user - - * config.c: when re-writing the config file, don't repeat the path - if it hasn't changed - - * config.c: when appending a list of variable to a category, ensure - the tail pointer points to the last variable in the list - - * config.c: clear the category's variable tail pointer as well when - variables are detached from it - - * config.c: ouch... don't use printf, use ast_log/ast_verbose - - * apps/app_voicemail.c, include/asterisk/config.h: 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 - -2006-11-15 15:17 +0000 [r47648-47655] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Send error message if we fail to allocate - sip socket, possibly caused by too few file handles (wasn't - possible before, but with the new way of sending temp messages, - it is). Found this bug under heavy load testing with SIPP. - - * channels/chan_sip.c: Sending 200 OK and not getting ACK is - considered critical for the call. - -2006-11-14 22:15 +0000 [r47631] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Update copyright information in the ADSI - logo blob. - -2006-11-14 11:06 +0000 [r47596] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Avoid collissions between the peerpoke - system and the retransmits. Issue #8272. In some cases, changed - timers caused the retransmit system to destroy the dialog before - peerpoke_expire got a chance. - -2006-11-13 21:26 +0000 [r47583] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_pgsql.c: Initialize global pointers for connection and - result to NULL. (issue #8356 reported by james) - -2006-11-13 20:18 +0000 [r47580] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Having more than 255 old messages caused - corruption in the new/old count - -2006-11-13 19:04 +0000 [r47571] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't send 487 if we've already sent 200 OK - on invite at time of receiving a BYE in the same transaction. - (SIPP testing) - -2006-11-13 17:05 +0000 [r47549] Joshua Colp <jcolp@digium.com> - - * apps/app_sms.c: When sending an SMS with a user data header - properly set the UDH flag in the first byte. (issue #8347 - reported by hoffmeis) - -2006-11-13 05:45 +0000 [r47522-47525] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: If the execute fails a second time, make sure - that we don't pass back a stale handle - - * channels/chan_zap.c: Don't play dialtone if the seizing the - channel fails (Bug 7754) - -2006-11-12 06:09 +0000 [r47496] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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) - -2006-11-10 20:46 +0000 [r47452-47470] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Clear dialog on loop (backport from 1.4 by - mistake) - - * channels/chan_sip.c: - Don't check for ignore in blocks that - isn't reached if ignore is on... - return properly after sending - reply in handle_request_invite - - * channels/chan_sip.c: Fix multipart/mixed SDP support (issue 8010, - alphaque) - -2006-11-09 16:48 +0000 [r47379] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c: 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 13:09 +0000 [r47359] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h: Fixed segfault when no - misdn.conf exists, reported by Igor Neves, thanks. - -2006-11-08 07:40 +0000 [r47307-47308] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Remove dialog properly at unload of module - (rizzo) - -2006-11-07 18:22 +0000 [r47274] Steve Murphy <murf@digium.com> - - * include/asterisk/channel.h, channel.c: This mod for bug_7506, to - make the manager code output the proper event - -2006-11-07 13:02 +0000 [r47248] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't ever reply to an ACK. (Issue 8265) - -2006-11-07 01:22 +0000 [r47238] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: 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) - -2006-11-02 17:47 +0000 [r46964] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: ignore files in a music on hold directory - that begin with '.' (issue #8249, cboie) - -2006-11-02 15:15 +0000 [r46899] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't overwrite flags in the packet - -2006-11-02 13:55 +0000 [r46876] Russell Bryant <russell@digium.com> - - * callerid.c: Add a missing call to free before returning in an - error condition (issue #8268, mrness) - -2006-11-01 21:20 +0000 [r46838] Matt O'Gorman <mogorman@digium.com> - - * logger.c: fix for bug #8083 crash caused by double free on m->msg - -2006-11-01 19:52 +0000 [r46803] Steve Murphy <murf@digium.com> - - * 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 18:24 +0000 [r46776] Russell Bryant <russell@digium.com> - - * res/res_monitor.c: 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) - -2006-10-31 15:46 +0000 [r46662] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_curl.c: 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 09:49 +0000 [r46585-46610] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Another try to fix - ;rport NAT traversal support (issue #7473) - - * channels/chan_sip.c: If peer fails ACL check, fail the REGISTER - attempt - - * channels/chan_sip.c: On the other hand, we already copy the NAT - flags... Reverting. - - * channels/chan_sip.c: Issue 7473 - support ;rport on REGISTER - requests too. - -2006-10-31 06:18 +0000 [r46557-46560] Russell Bryant <russell@digium.com> - - * utils.c: 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: fix some copy/paste bugs in the checking of - arguments for the "control stream file" AGI command (issue #8255, - mnicholson) - -2006-10-30 16:00 +0000 [r46402-46430] Olle Johansson <oej@edvina.net> - - * rtp.c: Bind rtcp to proper IP address - - * channels/chan_sip.c: Issue #7869 - Stop sending 302 redirect when - not getting an answer... - - * channels/chan_sip.c: issue #7608: Notifications with wrong - content-type. Reported by jsiddall. - -2006-10-27 17:36 +0000 [r46361] Russell Bryant <russell@digium.com> - - * res/res_agi.c, asterisk.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) - -2006-10-27 09:24 +0000 [r46350] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - 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 - -2006-10-26 20:06 +0000 [r46344] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7240, by mistake only committed to - trunk (now 1.4), reported by edgreenberg in Issue #7966. Thanks - Ed! - -2006-10-26 17:47 +0000 [r46332-46337] Jason Parker <jparker@digium.com> - - * contrib/scripts/astgenkey.8: oops - somebody forgot to change - this - long ago, probably. - - * channels/chan_skinny.c: Remove a useless ast_mutex_unlock. Issue - #8186, patch by anthonyl (fix suggested by benh). - -2006-10-25 19:28 +0000 [r46213-46258] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Working to resolve #7608 - adding debug - output - - * channels/chan_sip.c: Fix the attack shield for 1.2 too. REFER and - NOTIFY can create dialogs in the world of Asterisk. - -2006-10-25 08:41 +0000 [r46176] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - added nttimeout option to configure wether we disconnect calls on - NT timeouts or not during an overlapdial session - -2006-10-23 00:25 +0000 [r45927] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_odbc.c: Don't leak memory mmmk? - -2006-10-21 12:35 +0000 [r45808] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: 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. - -2006-10-19 17:16 +0000 [r45691] Joshua Colp <jcolp@digium.com> - - * apps/app_externalivr.c: Respect language selection when seeing if - the file exists (issue #8178 reported by mnicholson) - -2006-10-17 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.13 released - -2006-10-17 20:37 +0000 [r45380] Joshua Colp <jcolp@digium.com> - - * 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 17:50 +0000 [r45332] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an integer signedness problem. - -2006-10-17 17:22 +0000 [r45326] Kevin P. Fleming <kpfleming@digium.com> - - * LICENSE: provide licensing language for IAXy firmware file - -2006-10-17 15:50 +0000 [r45306] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: After some - research, we realized that the default behaviour since a long - time was doing the right thing, even though the change optimized - a bit and removed a lot of potential risks. Conclusion: No need - for a configuration option at all. - -2006-10-16 19:59 +0000 [r45260-45265] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: Use responses - rather then replies even though they mean the same thing. - - * channels/chan_sip.c, configs/sip.conf.sample: Add - 'ignoreoodreplies' option which will not create a pvt structure - on a SIP response but instead basically drop it. - -2006-10-14 00:16 +0000 [r45134] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: Made a small update to solve bug 8128; The - switch-case fallthru goto to a pattern extension needed to - resolved the wildcards to an appropriate digit for extension - matching to work - -2006-10-13 22:57 +0000 [r45119] Kevin P. Fleming <kpfleming@digium.com> - - * 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 19:27 +0000 [r45090] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: avoiding warning, fixing potential bug - (backported from 1.2) - -2006-10-13 17:01 +0000 [r45060] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c: Turn on volume adjustment if it needs to be - on (issue #8136 reported by mnicholson) - -2006-10-13 16:18 +0000 [r45048] Kevin P. Fleming <kpfleming@digium.com> - - * 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 15:49 +0000 [r45030] Joshua Colp <jcolp@digium.com> - - * 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 13:11 +0000 [r44993-45020] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.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 18:31 +0000 [r44955] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/utils.h, channels/chan_sip.c, utils.c, - netsock.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) - -2006-10-10 13:34 +0000 [r44785] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: (re)added - support of dynamical enabling hdlc on bchannels - -2006-10-09 14:36 +0000 [r44757] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8101 - wrong parameter for screening - in remote-party-id - -2006-10-06 16:52 +0000 [r44501-44580] Joshua Colp <jcolp@digium.com> - - * file.c: Even more frames to treat as though the remote side - disappeared (issue #8097 reported by eldadran) - - * file.c: Treat busy control frames as hangup in the file streaming - core (issue #8097 reported by eldadran) - -2006-10-05 10:02 +0000 [r44460] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed segfault which happens during - hold/transfer action - -2006-10-05 01:27 +0000 [r44392-44432] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix Polycom presence notification again - - * 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) - -2006-10-04 16:02 +0000 [r44343] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: For bug 7776, I have inserted a warning about - Macro nesting vs. stack limitations - -2006-10-04 15:26 +0000 [r44334-44335] Christian Richter <christian.richter@beronet.com> - - * 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_config.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, - configs/misdn.conf.sample, channels/misdn/isdn_lib.c: 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 - -2006-10-03 20:14 +0000 [r44296] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix a logic error in my previous fix to the - queue reload code - -2006-10-02 20:07 +0000 [r44168-44213] Joshua Colp <jcolp@digium.com> - - * 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) - - * 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 13:28 +0000 [r44149] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_lib.c: fixed the hold/retrieve/transfer - issues, removed a useless bc field, added setting of - frame.delivery fields, some minor code cleanups - -2006-10-01 15:19 +0000 [r44110] Russell Bryant <russell@digium.com> - - * configs/queues.conf.sample: Fix the name of the - "eventmemberstatus" option in the sample queues.conf (issue - #8065, adamg) - -2006-09-29 13:44 +0000 [r43977] Kevin P. Fleming <kpfleming@digium.com> - - * cli.c: proper fix for ast_group_t change - -2006-09-28 18:00 +0000 [r43924] Joshua Colp <jcolp@digium.com> - - * frame.c, include/asterisk/logger.h, channels/chan_misdn.c, - channels/chan_sip.c, channels/chan_skinny.c, - funcs/func_timeout.c, apps/app_festival.c, res/res_features.c, - apps/app_hasnewvoicemail.c, apps/app_alarmreceiver.c, - channels/iax2-provision.c, res/res_musiconhold.c, - res/res_monitor.c: Put in missing \ns on the end of ast_logs - (issue #7936 reported by wojtekka) - -2006-09-28 17:31 +0000 [r43916] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix buggy (and overly complex) loop used during - reload of app_queue for static member list updating - -2006-09-28 16:37 +0000 [r43897] BJ Weschke <bweschke@btwtech.com> - - * 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 16:32 +0000 [r43895] Kevin P. Fleming <kpfleming@digium.com> - - * cli.c: eliminate compiler warning introduced by recent changes - -2006-09-28 16:13 +0000 [r43891] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Stop the stream after waitstream returns so - that our formats get restored. (issue #7370 reported by - kryptolus) - -2006-09-28 15:18 +0000 [r43871] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Fix race condion crash with get_member_status - (#7864 - tim_ringenbach reported and patched) - -2006-09-27 20:20 +0000 [r43815] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Avoid inability to lock directory log - message by creating the directory ahead of time. (Issue 7631) - -2006-09-27 19:35 +0000 [r43800] Jason Parker <jparker@digium.com> - - * apps/app_playback.c, pbx.c: Playback() wasn't setting - PLAYBACKSTATUS under several circumstances. Playback() returns -1 - on missing args - so should Background() - -2006-09-27 16:54 +0000 [r43778] Russell Bryant <russell@digium.com> - - * res/res_features.c, channel.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 12:51 +0000 [r43764] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_lib.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. - -2006-09-26 20:49 +0000 [r43708] Russell Bryant <russell@digium.com> - - * 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-4) - -2006-09-26 20:38 +0000 [r43705-43706] Joshua Colp <jcolp@digium.com> - - * .cleancount: I changed the channel structure... let's be sure - this gets updated! - - * channels/chan_sip.c, include/asterisk/channel.h: Use proper type - to represent the group variable (issue #8025 reported by makoto) - -2006-09-26 20:23 +0000 [r43699] Russell Bryant <russell@digium.com> - - * 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) - -2006-09-25 21:14 +0000 [r43634] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue - 7824): 1) delete=yes was ignored 2) maxmessages was ignored - -2006-09-24 13:50 +0000 [r43552] Russell Bryant <russell@digium.com> - - * 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 21:53 +0000 [r43509] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c, channel.c: Yay another 'round of spy fixes! - This fixes a small logic flaw with the cleanup function and a - memory allocation issue. (issue #7960 reported by jojo & issue - #7999 reported by aster1) Special thanks to csum77 for letting me - into a box where this issue was happening. - -2006-09-21 17:01 +0000 [r43409-43420] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_rpt.c: Whitespace change... really just an excuse to - test repotools - - * cdr/cdr_tds.c, cdr/Makefile: TDS 0.64 updates - -2006-09-20 05:08 +0000 [r43314] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_misdn.c, channels/chan_sip.c, - channels/chan_skinny.c: make some more functions static - -2006-09-19 16:21 +0000 [r43269] Matt O'Gorman <mogorman@digium.com> - - * pbx/pbx_gtkconsole.c, apps/app_dial.c, channels/chan_sip.c, - apps/app_macro.c, asterisk.c, config.c, apps/app_queue.c, pbx.c: - fixes some verbose vs debug issues. patch from bug 2617 - -2006-09-19 12:28 +0000 [r43248] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: cid is passed to a destructive function; - thus a copy is needed (issue 7961) - -2006-09-18 20:08 +0000 [r43220] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7682 - don't add contacts to 4xx - responses. (Ugly fix, not proud at all) - -2006-09-18 15:30 +0000 [r43163] Joshua Colp <jcolp@digium.com> - - * apps/app_math.c: Add deprecation notice about app_math (issue - #7957 reported by k-egg) - -2006-09-18 15:05 +0000 [r43160] Steve Murphy <murf@digium.com> - - * configs/zapata.conf.sample: Clarified what "callwaiting" does in - zapata.conf. - -2006-09-18 15:05 +0000 [r43159] Joshua Colp <jcolp@digium.com> - - * configs/indications.conf.sample: Add number unobtainable tone for - New Zealand (issue #7969 reported by nic_bellamy) - -2006-09-17 13:54 +0000 [r43072] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Directory used the wrong context for - delivery of 0- and *- keypresses (according to Directory's own - documentation) - Issue 7965 - -2006-09-16 07:57 +0000 [r43003-43019] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_iax2.c: When a realtime peer expires, reset the - ipaddress in the realtime database back to 0 (Issue 6656) - - * apps/app_meetme.c: When the marked user enters the conference, we - should no longer timeout - -2006-09-14 22:16 +0000 [r42946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_zap.c: Error message references wrong argument - (Issue 7951) - -2006-09-13 19:51 +0000 [r42892] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Backport bugfix patch from 7918 to 1.2 - - msg_cfg destroyed before used - -2006-09-11 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.12.1 released - -2006-09-11 21:47 +0000 [r42697-42783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_meetme.c, apps/app_page.c: When paging, only wait 5 - seconds for the marked user to enter the conference. After that, - assume the paging already completed by the time the channel - entered the conference and drop back out. (Issue 7275) - - * configs/extensions.conf.sample, configs/alsa.conf.sample, - configs/zapata.conf.sample, configs/iax.conf.sample, - configs/osp.conf.sample, configs/dundi.conf.sample, - configs/enum.conf.sample, configs/vpb.conf.sample, - configs/cdr.conf.sample, configs/voicemail.conf.sample, - configs/phone.conf.sample, configs/misdn.conf.sample, - configs/sip.conf.sample, configs/skinny.conf.sample, - configs/features.conf.sample: Spelling/grammar fixes (Issue 7929) - - * configs/voicemail.conf.sample: Two grammar issues (bug 7927) - -2006-09-09 20:24 +0000 [r42600] Joshua Colp <jcolp@digium.com> - - * channel.c: Only truly consider the channel in the same format if - the format matches the raw format OR if a translation path - already exists to translate between them. (issue #7887 reported - by softins & issue #7803 reported by alvaro_palma_aste). Thanks - goes to stubert for giving me access to a box and showing me a - scenario where this occured. - -2006-09-09 12:14 +0000 [r42535] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Reset proper flag - Don't delete SIP - dialog prematurely Strangely enough imported from svn trunk... - It's confusing here in Greenland. (Committing from 36.000 feet - above Greenland, on the way to asterisk@von - http://www.pulver.com/asterisk ) - -2006-09-08 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.12 released - -2006-09-08 18:50 +0000 [r42452] Joshua Colp <jcolp@digium.com> - - * channel.c: Swap spies during masquerading - -2006-09-08 16:06 +0000 [r42421] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_authenticate.c: Jump logic was backwards: goto returns 0 - if it succeeds, and we should jump if authentication fails. (Bug - #7907) - -2006-09-08 04:37 +0000 [r42402] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: Use ast_best_codec to set the read/write - format - -2006-09-07 23:12 +0000 [r42355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_record.c: Format vulnerability fix - allowing the user - to specify a format is not a good idea (Bug 7811) - -2006-09-07 16:30 +0000 [r42260] Joshua Colp <jcolp@digium.com> - - * cdr.c: Let's use the same thing we use in other places to - calculate our time for ast_cond_timedwait (issue #7697 reported - by bn999) - -2006-09-07 02:14 +0000 [r42150-42200] Steve Murphy <murf@digium.com> - - * logger.c: This should fix the problem reported in 7564: logger - config file errors getting lost because logging isn't configured - yet. The problem was that the code that exists to handle this - case was not getting reached, because other tests were causing an - early return from ast_log(). - - * Makefile: added hours,minutes,seconds .gsm files to the install - portion of the makefile, as per bug 7545 - -2006-09-06 20:02 +0000 [r42148] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: Don't close the second file descriptor if it's the - same as the first one, as it will have already been closed - elsewhere and could cause massive panic. (issue #7699 reported by - bn999) - -2006-09-06 18:16 +0000 [r42133] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Look ma! No more deadlocks! <sic> As - posted from #7458 and others similar to it in Mantis: p->app_lock - was a mutex really designed for use with agents not in callback - mode. That being the case, I've tried to code it so that when - callback mode is used, the app_lock mutex will not be - locked/unlocked at all. Please let me know how you make out - and - if you continue to deadlock now, please reproduce the deadlock - logging information and post to Mantis. - -2006-09-06 17:10 +0000 [r42110] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed pipe consuming bug when using - chanIsAvail (#7878), also moved a debug log to the very begining - of misdn_hangup. - -2006-09-06 15:55 +0000 [r42054-42086] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Make realtime regseconds work as people - expected (0 on registration expiration or release, and actual on - normal state) (issue #7684 reported by kshumard) - - * include/asterisk/chanspy.h, apps/app_chanspy.c, - apps/app_mixmonitor.c, channel.c: Merge in last round of spy - fixes. This should hopefully eliminate all the issues people have - been seeing by distinctly separating what each component - (core/spy) is responsible for. Core is responsible for adding a - spy to a channel, feeding frames to the spy, removing the spy - from a channel, and telling the spy to stop. Spy is responsible - for reading frames in, and cleaning up after itself. - -2006-09-05 16:27 +0000 [r42014] Jason Parker <jparker@digium.com> - - * configs/zapata.conf.sample: Small typo in zapata.conf.sample - Reported by ppyy in 7881 - -2006-09-04 15:46 +0000 [r41989] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't kill the pvt before we have sent ACK - on CANCEL - -2006-09-03 17:38 +0000 [r41827-41882] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Make sure the forwarded channel inherits - variables appropriately when we receive a call forward in the - queue. (#7867 - raarts reported and patched) - - * apps/app_queue.c: Don't keep trying the same member in certain - strategies when members of the queue are unavailable (#7278 - - diLLec reported and patched) - - * apps/app_chanspy.c: Let's NOT spy on Zap/psuedo channels, - mmmmmmmmk? - - * apps/app_queue.c: Setting a retry of 0 is generally not a good - idea and shouldn't be allowed. (#7574 - reported by regin) - -2006-09-01 22:49 +0000 [r41768] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only wipe the redirected audio & video - IP/port if it's specified, and trigger a reinvite. - -2006-09-01 17:35 +0000 [r41716] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, include/asterisk/rtp.h, rtp.c: put in proper - fix for issue #7294 instead of the broken partial fix that was - committed, and thereby also fix issue #7438 - -2006-09-01 16:33 +0000 [r41690-41691] Joshua Colp <jcolp@digium.com> - - * channel.c: Finish up the last commit (was worse then originally - reported) - - * channel.c: Don't treat an unexpected control subclass as voice - (issue #7858 reported by PCadach) - -2006-08-30 19:01 +0000 [r41423] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7572 - Hangup when receiving a buggy - 487 response to an INVITE - -2006-08-30 18:59 +0000 [r41411] Russell Bryant <russell@digium.com> - - * channels/chan_mgcp.c, channels/chan_phone.c, - channels/chan_local.c, channels/chan_misdn.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_iax2.c: Restore original functionality of 1.2 in - places where ANI was not set, but was changed to be set. The - original change was done to ensure that the behavior of the - "callerid" option in each channel driver was consistent, but it - caused an unexpected behavior change of CDR records for users, so - this change is being reverted in 1.2. (issue #7695) - -2006-08-30 17:58 +0000 [r41390] Joshua Colp <jcolp@digium.com> - - * include/asterisk/lock.h: Properly handle an ETIMEDOUT result from - pthread_cond_timedwait (issue #7318 reported by arkadia) - -2006-08-30 14:31 +0000 [r41334] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 7822 - don't use SRV lookups if it's - disabled. - -2006-08-29 13:33 +0000 [r41269] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: clean up last commit ... most notably, there is - no reason to do heap allocations here, and it also included a - potential memory leak - -2006-08-29 05:49 +0000 [r41239-41262] Steve Murphy <murf@digium.com> - - * pbx/pbx_config.c: Fixes for bug 7813, via patch submitted by - stevens. - - * doc/README.variables: Removed from the docs the mention of the ! - and =~ operators, as these were knocked out of ast_expr2 because - they were new features. Let's hope I can keep them from getting - knocked out of the trunk, too! - - * apps/app_macro.c: According to a note added to 7731 by - mneuhauser, this will repair a break caused by the last fix - (7731). - -2006-08-25 15:21 +0000 [r41066-41069] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Don't send proceeding twice (#7800) - -2006-08-25 15:07 +0000 [r41065] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Text only - clarify the reason for entry - into authentication mode when the skipuser option is ignored - -2006-08-24 19:41 +0000 [r40994] Russell Bryant <russell@digium.com> - - * include/asterisk/linkedlists.h, channel.c, pbx.c: Fix a few - issues related to the handling of channel variables - in - pbx_builtin_serialize_variables(), the variable list traversal - would stop on a variables with empty name/values, which is not - appropriate - When removing the GROUP variables, use - AST_LIST_REMOVE_CURRENT instead of AST_LIST_REMOVE - During - masquerading, when copying the variables list from one channel to - the other, using AST_LIST_INSERT_TAIL is not valid for appending - a whole list. It leaves the tail pointer of the list invalid. - Introduce a new macro, AST_LIST_APPEND_LIST that appends a list - properly. (issue #7802, softins) - -2006-08-24 17:13 +0000 [r40971-40979] Joshua Colp <jcolp@digium.com> - - * configs/zapata.conf.sample: Minor documentation fix to add the - 'dynamic' dialplan option from angler - -2006-08-23 16:05 +0000 [r40901] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_agi.c: Revert last change - breaks retrieval of builtin - variables - -2006-08-22 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.11 released - -2006-08-22 02:59 +0000 [r40821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_random.c: Bug 7779 - Using initstate(3) means that we - cannot unload this module once loaded. - -2006-08-21 22:34 +0000 [r40798] Matt O'Gorman <mogorman@digium.com> - - * asterisk.c: Move the load_modules call so that if a module needs - realtime support it will work, none do currently but a good move - none the less. - -2006-08-20 22:09 +0000 [r40692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * CREDITS: Reformat to match the contribution style of other - contributors - -2006-08-20 04:49 +0000 [r40601] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Turn media level c= parsing on by default - (issue #7725 reported by psm) - -2006-08-19 01:03 +0000 [r40446] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, apps/app_directory.c: Fix a bug with - app_voicemail when trying to use app_directory to leave messages - to another user (options 3, 5, 2). If the context/extension - didn't exist in the dialplan (and why should it have to?), it - would fail, saying that it's an "invalid extension". Fix was - different in svn trunk. (issue BE-71) - -2006-08-18 19:10 +0000 [r40310-40392] Kevin P. Fleming <kpfleming@digium.com> - - * configs/zapata.conf.sample: make a feeble attempt to avoid the - 'how do I enable my hardware echo canceler' questions - - * channels/misdn_config.c (added), channels/chan_misdn_config.c - (removed): rename file per crichter's request - -2006-08-17 21:57 +0000 [r40306] Christian Richter <christian.richter@beronet.com> - - * doc/README.misdn, channels/misdn/mISDN.patch (removed), - channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/fac.c (added), channels/misdn/Makefile, - channels/misdn/chan_misdn_config.h, channels/misdn/ie.c, - channels/misdn/fac.h (added), channels/misdn/portinfo.c - (removed), channels/misdn/isdn_lib_intern.h, - channels/chan_misdn_config.c, channels/misdn/isdn_msg_parser.c, - configs/misdn.conf.sample, channels/Makefile, - channels/misdn/isdn_lib.c: This rather small ;-) commit merges - the changes from my team branch 0.3.0 into t he 1.2 branch. These - changes include the new mISDN mqueue interface which makes it - possible to compile chan_misdn against the current cvs version of - mISDN/mISDNuser. These changes also contain various additions and - numerous bugfixes to chan_misdn . Each change is documented in - the commit logs in the team/crichter/0.3.0 branch. - -2006-08-17 16:36 +0000 [r40227] Russell Bryant <russell@digium.com> - - * channel.c: revert bogus change to attempt to fix bug 7506 which - actually causes half of the channels not to get "Newchannel" - events at all (issue #7745) - -2006-08-17 16:22 +0000 [r40223-40225] Joshua Colp <jcolp@digium.com> - - * funcs/func_cdr.c: Use the last CDR entry instead of the first CDR - entry for variable retrieving variables using the CDR dialplan - function. (issue #7689 reported by voipgate) - - * apps/app_macro.c: Make app_macro compile again - -2006-08-17 16:07 +0000 [r40220] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: In app_macro, changed the previously changed - upper recursion depth limit to a variable, default of the - original val of 7. MACRO_RECURSION is a channel variable that - will override the limit, but until I can understand and fix why - this limit is neccessary, I am not advertising this variable in - the docs. This fix mirrors the changes made in r40200 in trunk. - -2006-08-16 18:57 +0000 [r40057] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_mgcp.c: don't allow AUEP responses to overflow the - stack during a string copy (reported by Mu Security) - -2006-08-15 22:49 +0000 [r39935] Russell Bryant <russell@digium.com> - - * res/res_agi.c: use pbx_builtin_getvar_helper() so that GET - VARIABLE can retrieve global variables (issue #7609) - -2006-08-15 22:13 +0000 [r39931] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: This revision fixes bug 7731, the inability for - macros to be called more than one level deep in the 'h' - extension. It also pushes up the limit of recursion depth from 7 - to 20. - -2006-08-08 18:39 +0000 [r39379] Kevin P. Fleming <kpfleming@digium.com> - - * CREDITS: add explicit listing of anthm's contributions (issue - #7683) - -2006-08-08 17:04 +0000 [r39350] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Increase the buffer size for the callid - (issue #7675, reported by pssatcs) - -2006-08-07 01:28 +0000 [r39081] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix a crash reported to me by hads on IRC. - This crash would occur with the use of the - "distinctiveringaftercid" option. Also, on this user's system, - the crash would only occur when built without optimizations. This - is because the bug is that the code would write past the end of - an array that was allocated on the stack, and the structure of - the stack is different with or without optimizations enabled. - -2006-08-07 00:15 +0000 [r39056] Joshua Colp <jcolp@digium.com> - - * channel.c: Reset our stream and vstream pointers back to NULL so - that any generator that uses them (file based MOH) will not try - to close them again. (issue #7668 reported by jmls) - -2006-08-05 09:01 +0000 [r38903-38982] Russell Bryant <russell@digium.com> - - * channel.c: Always generate a Newstate event in ast_setstate() - instead of making it a Newchannel event if the state was - AST_STATE_DOWN. The Newchannel event will always be generated in - ast_request(), so this just causes a duplicated Newchannel event - in some cases. (issue #7506, repoted by capouch, fixed by me) - - * apps/app_queue.c: remove duplicate queue log entry when the - caller exits on a timeout (issue #7616, ppyy) - - * channels/chan_sip.c: don't advertise that this function can set a - SIP header when it can only do reads - - * apps/app_dial.c: make sure the priv-callerintros directory exists - before trying to create a file there (issue #7659, patch by hads, - with some modifications by me) - - * channels/chan_mgcp.c, channels/chan_vpb.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_modem.c, channels/chan_iax2.c: Fix an issue that - would cause a NewCallerID manager event to be generated before - the channel's NewChannel event. This was due to a somewhat recent - change that included using ast_set_callerid() where it wasn't - before. This function should not be used in the channel driver - "new" functions. (issue #7654, fixed by me) Also, fix a couple - minor bugs in usecount handling. chan_iax2 could have increased - the usecount but then returned an error. The place where chan_sip - increased the usecount did not call ast_update_usecount() - - * channel.c: suppress a compiler warning about the usage of a - potentially uninitialized variable - -2006-08-03 19:54 +0000 [r38825] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Treat the file as invalid if we have no - valid formats for it (issue #7643 reported by KNK) - -2006-08-03 05:22 +0000 [r38761] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7648 - Checking wrong count for - plurality on new messages for Dutch language - -2006-08-02 19:29 +0000 [r38686-38731] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix brain-damage I introduced when trying to - fix the CANCEL/BYE sending mechanism for pending INVITES accept - unknown 1xx responses as 183 responses (as RFC3261 mandates we - should do) - - * res/res_features.c, channel.c: ensure that the 'feature digit - timeout' value is taken into account when deciding how long the - bridge should run (this fixes a problem report where a digit - press that did not invoke a feature is never passed across the - bridge) - -2006-08-01 19:20 +0000 [r38654] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Close the stream when file based MOH stop. - This won't get rid of their position in the file but it will - cause the translation path to be setup again. (issue #7634 - reported by asimpson) - -2006-07-31 21:14 +0000 [r38611] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't reissue hangup requests for SIP - channels that have expired their RTP timeouts (one time is - enough) don't rescan the SIP private structure list too fast, it - can cause channels to not be able to hang up (issue #7495, and - probably others) use ast_softhangup_nolock() since we already - hold the channel's lock - -2006-07-31 17:09 +0000 [r38585] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Add missing code to bring transferee channel - out of MOH/autoservice under certain circumstance (issue #7611 - reported by guillecabeza with minor mods by myself) - -2006-07-31 04:06 +0000 [r38546-38547] Russell Bryant <russell@digium.com> - - * frame.c: one more small tweak for thread-safety of TRACE_FRAMES - - * frame.c: Make the frame counting done with TRACE_FRAMES defined - thread-safe - -2006-07-29 23:18 +0000 [r38501] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: How many attempts does it take to make a SIP - URI parser that works well? I'm up to 5 personally. On to the - good stuff - parse the domain first, user second, and get rid of - port & options/params last. (issue #7616 reported by andrew) - -2006-07-28 18:49 +0000 [r38420] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make a copy of the request URI in - check_user_full instead of modifying the one on the structure, - and also strip params properly from the user portion of the SIP - URI so as to preserve the domain (issue #7552 reported by dan42) - -2006-07-27 22:23 +0000 [r38347-38370] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_chanspy.c: use the enum that defines the option - arguments, so that the likelihood of mismatched option indexes is - reduced (which in this case was a bug, the volume argument was - not checked properly) - - * channel.c: do a better job avoiding translation path - teardown/setup when not needed - -2006-07-27 04:25 +0000 [r38328] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix crash when using the "regexten" option - with MALLOC_DEBUG enabled. This was not reported in the bug - tracker but the same bug has been demonstrated in other places in - the code. - -2006-07-27 02:43 +0000 [r38310] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: don't do useless translation destroy/build when the - channel is already in the correct format - -2006-07-27 01:58 +0000 [r38288] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix a crash when MALLOC_DEBUG is enabled and - the regexten is enabled. The crash would occur when the extension - got removed. (fixes issue #7484) - -2006-07-26 15:26 +0000 [r38234] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Put default callerid into contact when the - one specified is either NULL or has a zero string length. (issue - #7590 reported by key2) - -2006-07-25 19:43 +0000 [r38200] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: This resolves a deadlock that a tech support - customer was getting frequently when his users would answer call - waiting. If another thread is currently holding the zt_pvt lock - for the first channel, unlock both channels and let asterisk - retry the native bridge, just like what is done for the second - channel directly below these changes. - -2006-07-24 17:05 +0000 [r38167] Steve Murphy <murf@digium.com> - - * codecs/gsm/Makefile: This fixes a compile problem for s390 as - reported in bug 7253. Tested on both an s390 and non-s390 - machine. - -2006-07-19 17:10 +0000 [r37949] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that global 'maxauthreq' is reset to - zero during 'reload' - -2006-07-18 00:41 +0000 [r37828-37856] Russell Bryant <russell@digium.com> - - * frame.c: don't crash if the frame has no data, but has a src - - * frame.c: if asked to duplicate a frame that has no data, don't - set the frame's data pointer past the end of the allocatted - buffer for the new frame - -2006-07-17 22:36 +0000 [r37765-37808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * formats/format_h263.c: Backport buffer increase to 1.2 - - * formats/format_h263.c: Overflow bad - -2006-07-15 23:29 +0000 [r37691] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * enum.c: Bug 7513 - ensure that each time we do a query, the - results are returned in the same logical order, so that when we - iterate over the list, we get all results, not some results - repeated, due to insufficient sorting. - -2006-07-14 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.10 released - -2006-07-14 13:31 +0000 [r37612] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_sms.c: Bug 7526 - previous commit broke app_sms - -2006-07-13 21:22 +0000 [r37571] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: don't fail/abort if the message category - sound file cannot be played, just generate a warning message and - continue message playback - -2006-07-13 18:44 +0000 [r37546] Russell Bryant <russell@digium.com> - - * rtp.c: yeah, ummm... This frame pointer should not be static. - This situation only exists in 1.2 (pointed out by Constantine - Filin on the asterisk-dev mailing list) - -2006-07-13 16:44 +0000 [r37531] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: report address of peer trying to subscribe - to unknown hint - -2006-07-13 15:45 +0000 [r37458-37516] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * doc/README.enum: Bug 7532 - Typo in enum example - - * contrib/init.d/rc.mandrake.zaptel: Merge fixup for asterisk - startup script to zaptel startup script - -2006-07-12 15:53 +0000 [r37441-37442] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: fix a weird case where a lock file could be - left (but would happen almost never) - - * app.c: fix a case where ast_lock_path() could leave a - randomly-named lock file hanging around make ast_unlock_path - actually report when unlocking fails - -2006-07-12 15:23 +0000 [r37439] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Add support to have maxauthreq as a global - option - -2006-07-12 13:54 +0000 [r37417-37419] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, utils.c, res/res_agi.c, apps/app_zapras.c, - asterisk.c, channels/chan_modem.c, channels/chan_iax2.c: remove - some more bad examples of using printf - - * enum.c, pbx/pbx_config.c: get rid of some more printf's (although - most of these were ifdef-ed out) - -2006-07-12 03:55 +0000 [r37402] Matt O'Gorman <mogorman@digium.com> - - * app.c: GRRR no fprintf! - -2006-07-11 19:00 +0000 [r37378] Joshua Colp <jcolp@digium.com> - - * configs/iax.conf.sample, channels/chan_iax2.c: Add configuration - option for IAX2 users that will limit the amount of outstanding - AUTHREQs we are waiting for replies on. - -2006-07-10 21:01 +0000 [r37361] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: do masquerade-behind-proxy checking with better - control over locks - -2006-07-07 23:57 +0000 [r37307] Joshua Colp <jcolp@digium.com> - - * rtp.c: Change message regarding marker bit forcing when SSRC - changes to be shown only during debug so it doesn't overload high - capacity systems - -2006-07-06 21:41 +0000 [r37224] Matt O'Gorman <mogorman@digium.com> - - * channel.c: patch resolves issue with when to decide if its right - time to native bridge, feature redirect was not being checked. - patch from bug #7296 - -2006-07-06 20:38 +0000 [r37212] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Don't do weird things on a callback agent - that has attempted logoff while still on the phone. - -2006-07-06 15:48 +0000 [r37173] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Instead of giving the scheduled item ID on a - peer expiration, give the time until they expire (issue #7455 - reported by slavon) - -2006-07-06 13:47 +0000 [r37143] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_db.c: Fix spelling/grammar (issue 7493) - -2006-07-05 15:31 +0000 [r36998] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c: Spell extension correctly in documentation - for chan_oss dial (issue #7487 reported by flefoll) - -2006-07-04 14:45 +0000 [r36838-36911] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Tell clients based on old SIP standard that - we only support MD5 digest authentication... - - * channels/chan_sip.c: issue #7470 - Need larger buffer for - record-route headers... - -2006-07-03 05:12 +0000 [r36697-36751] Russell Bryant <russell@digium.com> - - * asterisk.c: fix a race condition that caused asterisk to log a - *ton* of warnings on mac osx about poll returning an error - because the polled file descriptor was bad. - - * channels/chan_mgcp.c, channels/chan_phone.c, - channels/chan_local.c, channels/chan_misdn.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_agent.c, channels/chan_features.c, - channels/chan_h323.c, channels/chan_modem.c, - channels/chan_iax2.c: use ast_set_callerid to be more consistent - and to make sure that the "callerid" option in the conf files is - always handled the same way and sets ANI (issue #7285, gkloepfer) - - * dsp.c: fix the build with BUSYDETECT_TONEONLY defined (issue - #7414) - -2006-06-30 14:05 +0000 [r36290-36377] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Bug 7349 - Directory did not work correctly - when USE_ODBC_STORAGE was defined. - - * Makefile: Bug 7388 - compatibility changes for Solaris - -2006-06-29 07:19 +0000 [r36253-36254] Kevin P. Fleming <kpfleming@digium.com> - - * configs/queues.conf.sample: clarify documentation for - 'persistentmembers' setting - - * configs/sip.conf.sample: add documentation for peer-specific - 'outboundproxy' setting - -2006-06-28 14:12 +0000 [r36187] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't delete scheduled item twice in - sip_destroy (already fixed in svn trunk) - -2006-06-26 17:10 +0000 [r36078] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: ensure that two SIP channels that exist at - the same moment will not have the same channel names (issue - #7245, different fix) - -2006-06-26 15:27 +0000 [r36043] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6997 maybe, but anyway - don't - retransmit responses to NON-invite requests. - -2006-06-25 15:10 +0000 [r35915] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 7425 - Size of buffer is passed in by - len - -2006-06-23 11:30 +0000 [r35669] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: We should lock the queue before we go making - changes to member interface statuses. - -2006-06-21 19:25 +0000 [r35334] Joshua Colp <jcolp@digium.com> - - * configs/indications.conf.sample: Add Venezuelan indications - (issue #7402 reported by palillo) - -2006-06-20 15:05 +0000 [r35121] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * stdtime/private.h: Bug 7398 - Solaris puts its zoneinfo files in - a nonstandard place - -2006-06-20 10:27 +0000 [r35058] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6820 - Possible fix (already - implemented in trunk) - -2006-06-19 20:27 +0000 [r34911] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Call reset_user_pw upon changing the - password using externpass (issue #7395 reported by Ryan Cumming) - -2006-06-19 18:07 +0000 [r34875] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Issue 7357 - txt file left behind when - going to operator. Also, fix a possible file descriptor leak. - -2006-06-18 21:03 +0000 [r34627-34655] Russell Bryant <russell@digium.com> - - * pbx.c: don't set state to BUSY if the channel is already in the - UP state (issue #7376, backported from trunk) - - * configs/iax.conf.sample, channels/chan_iax2.c: don't store - multiple secrets delimited with semicolons for peers because this - is only valid for users. Instead, only keep the last specified - secret for a peer entry. Also, document how multiple secrets are - handled in the sample config. (Reported by PCadach on - #asterisk-bugs) - -2006-06-16 03:37 +0000 [r34400] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Zero out a declared structure so as to not - crash if it contains invalid data (reported by Qwell on - #asterisk-dev) - -2006-06-15 14:11 +0000 [r34306] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 7294 - patch by phsultan - Asterisk - sends Invite instead of BYE in some cases. - -2006-06-15 13:30 +0000 [r34274] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: don't use prefixed structure names for internal - structures don't use a plural structure name for a singular - object - -2006-06-15 12:40 +0000 [r34242] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: VoicemailMain exits on any key, when the - language is set to Italian, instead of properly handling the key - (issue 7353). - -2006-06-14 22:22 +0000 [r33841-34160] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: coding style cleanups on queue interface - handling code that was committed for the last release - - * channels/chan_iax2.c: use existing dial string parser for strings - supplied to iax2_devicestate, because they can be complete dial - strings, not just device names - - * include/asterisk/plc.h, jitterbuf.c, plc.c, apps/app_dumpchan.c, - apps/app_chanspy.c: clarify file headers that mention disclaimer - usage - - * file.c: don't output 'no format found' when we _did_ find the - format but couldn't open the desired file for some other reason - - * apps/app_mixmonitor.c: memory allocation optimizations - -2006-06-13 12:40 +0000 [r33753-33813] Russell Bryant <russell@digium.com> - - * pbx.c: remove duplicate mutex_unlock - - * apps/app_voicemail.c: fix various places where the code returns - without unlocking vmlock or destroying loaded configuration - - * apps/app_festival.c: add a missing close of an open fd, destroy - of open config, and removal of the calling channel from the - localusers list - - * asterisk.c: revert a change that caused more problems than it - fixed and fix the real problem in this code. fds was declared as - an array of zero size which caused some weird problems, some of - which would only be seen when compiling without optimizations. - (fixes issues #7071, #7326, and #7305) - -2006-06-12 21:34 +0000 [r33724] Joshua Colp <jcolp@digium.com> - - * include/asterisk/chanspy.h, apps/app_mixmonitor.c, channel.c: - Greatly simply the mixmonitor thread, and move channel reference - directly to spy structure so that the core can modify it. - -2006-06-12 20:40 +0000 [r33693] Russell Bryant <russell@digium.com> - - * res/res_agi.c: fix a place where a frame would be free'd twice - -2006-06-12 16:03 +0000 [r33638] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_local.c: only allow chan_local to masquerade the - outbound channel onto its owner, instead of the other way around - (this will ensure that group variables on the outbound channel are - preserved) - -2006-06-12 15:27 +0000 [r33615] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_agi.c: Move set priority up, because at this point in the - code, stdout is no longer the console. If we're unable to set - priority, the error goes to Asterisk as if it were an AGI command - (issue 7335). - -2006-06-11 21:21 +0000 [r33449-33548] Russell Bryant <russell@digium.com> - - * pbx.c: fix another place where a frame does not get free'd - - * apps/app_meetme.c: fix up five little places where frames would - not be free'd and remove an unnecessary mutex_unlock where there - is no way for it to be locked at that time - - * apps/app_ices.c: fix a place that would leak a frame (all of - these fixes are in applications that call ast_read() on a channel - but have code paths in them that would not free the frame) - - * apps/app_festival.c: fix a couple places that would leak a frame - - * apps/app_alarmreceiver.c: fix two places that would cause a frame - to be leaked - - * apps/app_url.c: fix a case where an HTML frame would be leaked - - * apps/app_test.c: Free frames read from the channel when measuring - noise. This resulted in about 9 or 10 seconds of leaked frames in - both the TestClient and TestServer applications - - * apps/app_zapbarge.c, apps/app_zapscan.c: backport a couple of - frame leak fixes from the trunk (revisions 33446, 33447) - -2006-06-09 18:52 +0000 [r33264-33300] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Allow the format outputted by meetme list to - be used for meetme commands (like kick) (issue #7322 reported by - darkskiez) - - * channels/chan_iax2.c: Remove an unneeded double lock (issue #7310 - reported by arkadia) - - * apps/app_dial.c: Handle hangup during recording of screened name - (issue #7304 reported by kulldominique) - - * apps/app_meetme.c: Add missing newlines (issue #7323 reported by - darkskiez) - -2006-06-09 15:53 +0000 [r33235] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Do not require a context on a domain= - setting - -2006-06-08 16:57 +0000 [r33036] Kevin P. Fleming <kpfleming@digium.com> - - * frame.c: handle out-of-memory conditions properly in - ast_frisolate() (reported by Slav Kenov on asterisk-dev mailing - list) - -2006-06-07 17:53 +0000 [r32818] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix some broken code with - BRIDGE_OPTIMIZATION defined (issue #7292) - -2006-06-06 16:55 +0000 [r32605] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7287 - A too short voicemail with - ODBC_STORAGE will cause the first voicemail to be deleted - erroneously - -2006-06-06 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.9.1 released - -2006-06-06 16:02 +0000 [r32582] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * callerid.c: Bug 7268 - Callerid leaks memory on error - -2006-06-06 15:48 +0000 [r32566] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: clean up yesterday's security fix to not - cause breakage when video mini frames are received - -2006-06-03 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.9 released - -2006-06-05 19:53 +0000 [r32373] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that the received number of bytes is - included in all IAX2 incoming frame analysis checks (fixes a - known vulnerability) - -2006-06-04 03:43 +0000 [r31921] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: return bridge exit logic to what it was before - i broke it :-( - -2006-06-03 17:02 +0000 [r31775] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: when using moh files mode, don't look for - a file past the number of files that have been loaded, or worse, - past the size of the files array - -2006-06-01 21:46 +0000 [r31321-31555] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_musiconhold.c: remove pointless forcing of the channel - into SLINEAR mode; the write format will be set later based on - the file that is chosen to be played to the channel - - * include/asterisk/channel.h, channel.c: handle Zap transfers - behind chan_agent properly so the agent doesn't get stuck waiting - for the call to hang up - - * configs/sip.conf.sample: remove a sample entry that never should - have been added (code to support it was not merged) - -2006-05-31 23:50 +0000 [r31194] Russell Bryant <russell@digium.com> - - * res/res_agi.c: if the connection to a FastAGI server fails - because of a timeout, log a more informative log message - -2006-05-31 22:26 +0000 [r31161] Kevin P. Fleming <kpfleming@digium.com> - - * rtp.c: silence a warning message that is not a warning - -2006-05-31 20:26 +0000 [r31127] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: fix misplaced manager event (issue #6866, - flefoll) - -2006-05-30 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.8 released - -2006-05-30 14:55 +0000 [r30770] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Fix infinite loop scenario and add some sanity - checking to prevent segfault on a NULL parameter coming in (which - probably shouldn't happen, but just to be safe...) - -2006-05-26 17:09 +0000 [r30424-30546] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: A new way to try and deal with deadlocks that - occur in app_queue at present. Using this approach, we only - manipulate the main queue mutexes when we get a dev state change - on a device that is actually a member of a queue. Backported from - /trunk for the "bug fix". - -2006-05-25 20:03 +0000 [r30373] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Don't play the enter sound twice when a person - joins a conference after the leader has joined it. (issue #6138 - reported by shanermn) - -2006-05-25 17:39 +0000 [r30293-30296] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/gsm/Makefile: don't try to use -march=s390 when building - on S/390 systems (reported via asterisk-users mailing list) - - * channels/chan_sip.c: allow SIPCHANINFO(peername) to work for - calls from users as well (issue #7215) - -2006-05-25 15:27 +0000 [r30239] Joshua Colp <jcolp@digium.com> - - * configs/extensions.conf.sample: Get rid of an incorrect SIP dial - string in the sample extensions.conf - I even tried variations... - no go (issue #7222 reported by arkadia) - -2006-05-24 21:24 +0000 [r30069-30098] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: oops... make sure to stop processing a - request once we have sent an authentication challenge (issue - #7220) - - * channels/chan_sip.c: don't send CANCEL on a pending INVITE if we - haven't received a provisional response yet... mark it pending - until the first response is received (issue #7079) - -2006-05-24 19:55 +0000 [r30037] Matt O'Gorman <mogorman@digium.com> - - * apps/app_meetme.c: app_meetme used the ast_max_exten instead of - path_max solves bug 6822 - -2006-05-24 19:44 +0000 [r30033-30035] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Merge branch for bug 6264 (Privacy option 2 - returns dial-status ANSWER / option_priority_jumping not - respected) (reported by jkoopmann and branch by murf) - - * logger.c: Fix deadlock caused by a race condition in the logger - when reloading (issue #7195 reported and fixed by softins) - -2006-05-24 16:59 +0000 [r29904-29973] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_agi.c: support video recording via AGI 'RECORD FILE' - command (issue #7068) - - * apps/app_queue.c: fix various bugs related to exiting from queue - via keypress and moh handling (issue #6776, different fix) - - * channels/chan_zap.c: respect 'usecallingpres' in zapata.conf even - if CLID has not been set for the channel (issue #7123) - - * channels/chan_sip.c, configs/sip.conf.sample: add an option to - allow the admin to 'hide' SIP user/peer names from systems trying - to 'fish' names - -2006-05-23 21:44 +0000 [r29849] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix the sourceaddress option (issue #7213, - alphaque) - -2006-05-23 18:16 +0000 [r29764] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: simplify/fix lock retry, and fix comment - -2006-05-23 17:17 +0000 [r29733] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: Sanity check code for an extended failure in - trying to obtain a channel lock that may have been obtained - elsewhere. Prevents the monitor thread of the SIP module from - going into an infinite loop, effectively, breaking SIP until you - restart Asterisk or the mutex is unlocked, whichever comes first. - -2006-05-23 17:15 +0000 [r29732] Kevin P. Fleming <kpfleming@digium.com> - - * dnsmgr.c, res/res_features.c, include/asterisk/linkedlists.h, - include/asterisk/lock.h, apps/app_sql_postgres.c, pbx.c: backport - some mutex initialization and linked list handling fixes from - trunk - -2006-05-23 15:58 +0000 [r29696] BJ Weschke <bweschke@btwtech.com> - - * res/res_features.c: Fix a potential leak and correct (hopefully) - a segfault under certain conditions. #6784 (vovan and perry - testing) - -2006-05-22 21:27 +0000 [r29464-29555] Joshua Colp <jcolp@digium.com> - - * apps/app_waitforsilence.c: Increase the silence threshold to 128 - to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed - by casper) - - * res/res_features.c: Use the correct language when playing the - transfer sound (issue #7109 reported by casper) - - * channels/chan_local.c: Preserve presentation bit when going - through chan_local (issue #7002 reported by acunningham) - -2006-05-22 14:59 +0000 [r29394-29398] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_meetme.c: Bug 7194 - spelling fix - - * pbx.c: Bug 7196 - month range did not work - -2006-05-21 15:16 +0000 [r29196] BJ Weschke <bweschke@btwtech.com> - - * res/res_features.c: When an application that is executed via - applicationmap and exits non-zero, make sure that we pass through - the correct return value from the application to make sure a - segfault doesn't occur by a bridge trying to continue when it - should not. Also, when executing applications via applicationmap, - make sure that the application is executed against the channel - whose DTMF caused it to be fired off in the first place. (part - 1/2 of #7090 - this is the only fix that will be applied to both - 1.2 and /trunk) acunningham and blitzrage on testing... - -2006-05-20 19:50 +0000 [r29052] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix the possibility of writing one byte past - the end of a buffer. (issue #7189, Mithraen) - -2006-05-20 02:35 +0000 [r28968] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: don't allow queue member devices to ring longer - than the total queue timeout (issue #6423, reported and patched - by bcnit) - -2006-05-20 02:31 +0000 [r28966] Russell Bryant <russell@digium.com> - - * apps/app_sms.c: fix a case where code made assumptions about how - memory for variables is allocatted on the stack - this patch is - slightly different than the one that went in for the trunk - -2006-05-20 00:55 +0000 [r28794-28896] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: don't try to predict where the compiler - will place things on the stack... put them in the right place - explicitly (issues #7029 and #7100, maybe others) - - * channels/chan_sip.c: use the specified 'subscribecontext' for a - peer rather than the context found via the target domain (domain - contexts are for calls, not for subscriptions) (issue #7122, - reported by raarts) - -2006-05-19 19:18 +0000 [r28754-28790] Russell Bryant <russell@digium.com> - - * utils/smsq.c: fix the build of smsq with -Werror. I learned - something new about format strings from this patch! (issue #7141, - Mithraen) - - * asterisk.c: This explicit poll is only needed on mac. In fact, it - breaks some systems such as some versions of Fedora, causing - 'asterisk -rx' to never exit. This has been tested on systems - showing the asterisk -rx problem, as well as other unaffected - versions of linux, mac osx 10.4, and FreeBSD 6. (issue #7071) - -2006-05-19 17:04 +0000 [r28627-28698] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Make the minidle option actually exist as - documented (issue #7159 reported by imran) - - * apps/app_voicemail.c: When forwarding messages use the context - that the active voicemail user was found in. (issue #7010) - - * enum.c: Backport of fix for issue #6654 that was fixed in trunk - but not here - - * apps/app_queue.c: Treat paused queue members as unreachable - (issue #7127 reported by peterh) - -2006-05-18 20:43 +0000 [r28335-28384] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix up a few more places to find the SDP - properly (fallout from fix for #7124) - - * channels/chan_sip.c: handle incoming multipart/mixed message - bodies in SIP and find the SDP, if present (issue #7124 reported - and patched by eborgstrom, but very different fix) - - * enum.c: use unsigned counters for handling answer/IE lengths - while processing DNS results (issue #7174) - - * channels/chan_sip.c: support 'inactive' tag for SDP media streams - (simple fix, proper fix will appear in 1.4 release) (issue #7130) - -2006-05-18 17:27 +0000 [r28257] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_hasnewvoicemail.c: Bug 7167 - HasNewVoicemail and - VMCOUNT() didn't work when USE_ODBC_STORAGE was defined - -2006-05-18 16:31 +0000 [r28169-28212] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Return -1 on error in ODBC messagecount and - 0 on success (issue #7133 reported by cfieldmtm) - - * apps/app_voicemail.c: Fix endless looping message by checking - value of res before doing retries stuff. (issue #7140 reported by - tanischen) - -2006-05-18 12:13 +0000 [r28125] Olle Johansson <oej@edvina.net> - - * apps/app_meetme.c: Video in meetme? Hmmm. Removed until we do - have some code for it. - -2006-05-17 22:34 +0000 [r27973] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Fix codec priority stuff during - authentication (issue #6194 reported by jkoopmann) - -2006-05-17 19:27 +0000 [r27927] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7176 - Crash in expire_register (We - need to find out what's causing peer to be undefined, so this is - just a bandaid, not a real fix) - -2006-05-17 17:07 +0000 [r27767-27847] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Priority jumping not working on VoiceMail - app with new syntax (issue #7164 reported and fixed by - alvaro_palma_aste) - - * apps/app_osplookup.c: OSPNext does not handle success/failure - correctly (issue #7147 reported and fixed by eborgstrom) - -2006-05-17 09:21 +0000 [r27723] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: chan_sip did not use the TRANSFER_CONTEXT - for transfers, like res_features. Now fixed. - -2006-05-17 02:19 +0000 [r27636] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7125 - Fix race condition between - resequencing and leaving a message - -2006-05-16 23:31 +0000 [r27594] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Inherit channel variables during call forwards - when going through chan_local (issue #7095 reported by raarts) - -2006-05-16 20:05 +0000 [r27468] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: don't leak frames when deferring DTMF or dropping - duplicate ANSWER frames (issue #7041, slightly different fix, - reported/patched by clausf) - -2006-05-13 04:08 +0000 [r27093] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7134 - File descriptor leak with ODBC - storage of voicemail - -2006-05-11 23:02 +0000 [r27051] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_logic.c: Bug 7086 - pbx_checkcondition substitution, - so that arbitrary strings are true (for regex) - -2006-05-11 09:05 +0000 [r26760-26773] Kevin P. Fleming <kpfleming@digium.com> - - * rtp.c: backport fix from trunk for bug #6934, ensuring that RTP - mark bit is changed when SSRC changes - - * channels/chan_sip.c: ensure that we send a response to REGISTER - requests that are successfully authenticated but contain invalid - Contact URIs - -2006-05-09 14:18 +0000 [r26050-26090] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c, doc/README.variables: Add the appropriate - jumping behavior that is the standard for 1.2.X to SIPGetHeader - that is now deprecated in /trunk. #7111 (blitzrage!!!) - - * apps/app_voicemail.c: Correct memory leak in find_user_realtime - #7118 (fnordian) - -2006-05-08 15:09 +0000 [r25608] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 7103 - mikma - The header is named - "Require" - Don't reply to ACK (Not using patch against trunk) - -2006-05-08 14:12 +0000 [r25518-25563] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Don't show agents as available when they - are in wrap-up time. #6726 (ZX81) - - * apps/app_queue.c: Make QueueStatusComplete event thread safe by - wrapping it inside the queue lock clause already there. #7013 - (bziherl reporting) - - * apps/app_queue.c: Don't recheck valid_exit() after getting the - result from say_position (which already checks it). Should - prevent another loop if the caller hits digits during the - position announcement. #6776 (tgj reporting) - -2006-05-08 11:16 +0000 [r25442] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Incorrect log statement when playing transfer - sounds (issue #7008 reported and fixed by nathan) - -2006-05-07 13:38 +0000 [r25288-25322] BJ Weschke <bweschke@btwtech.com> - - * apps/app_meetme.c: Fix playback behavior to exit correctly when - we receive a hangup during playback of the invalid pin message. - #7091 (AntD reporting) - - * asterisk.c: Reset the value of ast_mainpid if we fork so future - remote unix connections display the correct PID. #7098 (tzafrir - reporting) - -2006-05-06 02:32 +0000 [r25015-25165] Russell Bryant <russell@digium.com> - - * frame.c: fix a problem where the frame's data pointer is - overwritten by the newly allocated data buffer before the data - can be copied from it. This is in the ast_frisolate() function - which is rarely used. (issue #6732, stefankroon) - - * channels/chan_zap.c: ensure that the appropriate manager events - are sent in all of the places where alarms are detected or - cleared (issue #6866, flefoll) - - * channels/chan_h323.c: update chan_h323 to reflect the new - prototype for rtp_set_peer (issue #6560, casper) This was fixed a - couple months ago in the trunk, but never in 1.2. - -2006-05-05 20:44 +0000 [r25014] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c, include/asterisk/app.h, app.c: Voicemail - fixes along with an API change approved by russellb to fix the - bug(s). (jcollie and supczinskib) #7064 - -2006-05-05 17:39 +0000 [r24837-24911] Russell Bryant <russell@digium.com> - - * apps/app_while.c, apps/app_macro.c: use pbx_checkcondition() - instead of ast_true() to evaluate the condition for MacroIf and - WhileIf (issue #7086) - -2006-05-04 16:27 +0000 [r24706] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_queue.c: Bug 7023 - reload should not unpause members - -2006-05-04 11:17 +0000 [r24567-24669] BJ Weschke <bweschke@btwtech.com> - - * apps/app_verbose.c: Make sure that only the "|" is a recognized - delimiter for Verbose(), as the app documentation already - specifies. #7080 (alessiof reporting) - - * apps/app_dial.c: Correct application documentation to make users - aware that certain options cannot be used in conjunction with - others. #6666 (chotaire) - -2006-05-03 18:31 +0000 [r24496] Russell Bryant <russell@digium.com> - - * redhat/asterisk.spec: fix up "make rpm" - don't reference the - gzipped man page, because we don't store them compressed anymore - - add some files that currently were not listed (issue #6837) - -2006-05-03 12:39 +0000 [r24381] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7074 - Problem with long contact - lines - -2006-05-02 19:39 +0000 [r24295] BJ Weschke <bweschke@btwtech.com> - - * file.c: Make certain ast_stopstream() sets the channel's stream - members to NULL after closing them. #7067 (jcomellas) - -2006-05-02 02:12 +0000 [r24019-24097] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_privacy.c: Prompt does not request '#' to end input, so - the application should not require it - - * apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c, - apps/app_zapras.c, asterisk.c, apps/app_externalivr.c, - apps/app_ices.c, res/res_musiconhold.c, - include/asterisk/options.h: Bug 6864 - drop realtime priority on - ALL external processes - -2006-05-01 19:34 +0000 [r23985-23988] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c: Make sure that when someone 0's out while - recording a msg and then chooses to DELETE the recorded file, the - .txt file isn't left around by itself to cause problems later. - #7061 (dimitripietro reporting, blitzrage confirmed) - -2006-05-01 15:12 +0000 [r23951] Russell Bryant <russell@digium.com> - - * pbx.c: add missing locking of the dialplan functions list in the - "show functions" CLI command - -2006-05-01 10:45 +0000 [r23305-23899] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_skel.c: fix this to actually compile so people can learn - from it - - * cdr/cdr_sqlite.c: eliminate compiler warning - - * channels/chan_iax2.c: remove a pointless comparison, since the - buffer is smaller than the length being checked for - - * Makefile, editline/configure, cdr/Makefile, channels/Makefile, - db1-ast/Makefile: allow top-level OPTIMIZE setting to affect - builds in these subdirectories too - - * Makefile: let the compiler determine whether hardware or software - floating point should be used (like we do in the editline - subdirectory) - - * Makefile, apps/Makefile: remove extraneous -m64 flag that is not - needed remove old 'look' target which is no longer needed (these - are coming from Debian patches <G>) - - * editline/makelist: ensure that the script output is correctly - generated when the system's character set does not use the - English lowercase/uppercase character groups - - * Makefile: do installation in subdirs as a separate target (so - external modules can use the Makefile more easily) generate final - messages -after- running any post-install script that may be - present - -2006-04-28 16:40 +0000 [r23176] Russell Bryant <russell@digium.com> - - * configs/zapata.conf.sample, configs/mgcp.conf.sample, - configs/sip.conf.sample: note that group assignments must be from - 0 to 63 (issue #7048) - -2006-04-27 19:11 +0000 [r22954] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Queue(somequeue,,,,) -> interpreted as - Queue(somequeue,,,,0) (issue #7044 reported nathan fixed by - jsmith - sort of) - -2006-04-27 16:12 +0000 [r22866] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Fix buglet in channel reassignment on - SETUP_ACK - -2006-04-26 19:18 +0000 [r22596] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: do not allow for users to forward voicemail - to themselves, patch from 7001 - -2006-04-21 22:39 +0000 [r22112-22113] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channel.c: Bug 7004 - release all threads waiting on a condition - prior to freeing it - -2006-04-19 21:10 +0000 [r21638] Kevin P. Fleming <kpfleming@digium.com> - - * contrib/scripts/safe_asterisk.8, contrib/scripts/safe_asterisk: - support system-specific scripts in safe_asterisk, before starting - Asterisk proper - -2006-04-19 18:43 +0000 [r21597] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_odbc.c: Bug 6553 - plug memory leaks when ODBC connection - is down - -2006-04-18 23:31 +0000 [r21237] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: properly handle brace-wrapped strings in variable/function - references in the dialplan - -2006-04-18 06:26 +0000 [r20966-21037] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_random.c: Bug 6984 - off by one error in Random() - - * res/res_musiconhold.c: Bug 6544 - when we remove a music class, - the thread servicing it should die - -2006-04-14 17:21 +0000 [r20034-20037] Kevin P. Fleming <kpfleming@digium.com> - - * sounds.txt: uncomment files that actually do exist (oops) - - * sounds.txt: update text to match actual prompts being distributed - (thanks to Kinsey in the support department for reviewing all the - prompts!) - -2006-04-13 20:37 +0000 [r19891] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6947 - Allow vm broadcasts to more than - 256 characters worth of mailboxes - -2006-04-13 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.7.1 released - -2006-04-13 17:40 +0000 [r19812] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_page.c: oops... let's not set a variable and then - immediately overwrite it while assuming its old value will - magically return - -2006-04-13 15:56 +0000 [r19768] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx.c: Bug 6957 - variable names beginning with CALLERID weren't - substituted correctly - -2006-04-12 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.7 released - -2006-04-11 22:39 +0000 [r19394-19397] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c: Bug 6490 - telco intercept should report - NOANSWER instead of CHANUNAVAIL - - * apps/app_voicemail.c: Bug 6061 - Fix ODBC storage of VM on PGSQL - and MSSQL - -2006-04-11 21:58 +0000 [r19353] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't create a 'voicemail' symlink in the sounds - directory; app_voicemail has not needed it since January of 2005 - (issue #6613) - -2006-04-11 21:55 +0000 [r19351] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * asterisk.c: Bug 6097 - possible descriptor leak - -2006-04-11 21:50 +0000 [r19345-19348] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_page.c: don't call the originating device as part of the - Page() operation (issue #6932) - - * channel.c: simplify spy queue flushing logic, and always force a - flush when one side gets full, even if the other side is not - empty (issue #6457) - - * pbx/pbx_config.c: don't destroy the entire dialplan during - 'reload', just atomically replace it like 'extensions reload' - does (issue #6047) - -2006-04-11 20:46 +0000 [r19303] Joshua Colp <joshnet@nbnet.nb.ca> - - * include/asterisk/linkedlists.h: Minor linked lists bug fix. When - you're dealing with swapping entries around a lot it can cause a - seg fault. - -2006-04-11 20:11 +0000 [r19301] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c: handle call time limit properly when warning is - requested _after_ call would hae already ended (issue #6356) - -2006-04-11 01:05 +0000 [r18866-19008] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c, app.c: When using the silence detector in - ast_play_and_record() and ast_play_and_prepend(), the truncation - code never gets called to remove the detected silence, because - the value of res is zero when control gets to that point. #6903 - w/some mods (softins) - - * res/res_features.c: Don't say that we can pass an 'exten' - argument in the documentation of Park() when we really cannot. - #6902 (opsys) - -2006-04-08 19:20 +0000 [r18436-18494] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6914 - .txt file fails to rename on - operator out - - * formats/format_jpeg.c: Bug 6913 - fix for possible buffer - overflow - -2006-04-07 14:16 +0000 [r18250-18260] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Fix cause codes - Add cause code for - incompatible formats - - * channels/chan_sip.c: - Fix possible minor memory leak in chan_sip - - Return proper cause code on memory allocation error - -2006-04-06 22:15 +0000 [r18087-18089] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_meetme.c: fix typo - - * apps/app_meetme.c: small fix... don't try to check conference - details if it couldn't be created or found - - * apps/app_meetme.c: don't try to support 'i' or 'r' options if - chan_zap is not loaded, and warn the user when they attempt to - use them (issue #6675) update application help text to more - clearly define when Zaptel and chan_zap are required - -2006-04-06 17:24 +0000 [r17945] Russell Bryant <russell@digium.com> - - * apps/app_alarmreceiver.c: move continue out of block that checks - verbose level (issue #6880) - -2006-04-06 17:00 +0000 [r17702-17905] Joshua Colp <joshnet@nbnet.nb.ca> - - * pbx.c: Unlock channel on failure so that ast_mutex_destroy - doesn't throw a fit (issue #6647 reported by casper) - -2006-04-05 06:50 +0000 [r17335-17489] Olle Johansson <oej@edvina.net> - - * CREDITS, enum.c: Issue #6654: Enum crash on ADDRESS record, - possibly bad record, but still a crash - - * channels/chan_zap.c: Issue #6878 - Unhide DNDstate manager events - (thanks casper) - - * apps/app_queue.c: Issue #6882 - move "res=-1" out of verbosity - block, minor code cleanups (casper) - -2006-04-04 15:24 +0000 [r17283] Matt O'Gorman <mogorman@digium.com> - - * apps/app_senddtmf.c: Adds documentation to show what the w flag. - Patch from Ian Kinner at Digium. - -2006-04-03 20:38 +0000 [r17074-17150] Olle Johansson <oej@edvina.net> - - * configs/features.conf.sample: Issue 6870 - document that parking - lots need to be numeric - - * channels/chan_sip.c: Issue #6848 take two - Use the tag provided - by the SUBSCRIBE request when sending NOTIFY - - * channels/chan_sip.c: Ugly patch to avoid hangup causes in - non-final responses - -2006-03-31 19:11 +0000 [r16744-16771] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: move a NULL check to before the first time - the pointer is dereferenced (issue #6832) - - * channels/chan_iax2.c: fix the situation where bindport is - specified but bindaddr is not (issue #6616) - -2006-03-31 18:24 +0000 [r16742] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: ensure that hint watchers (subscribers) cannot be added or - removed while the dialplan is being modified - -2006-03-30 22:56 +0000 [r16579-16581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 6853 - Manager fixes: 1) extra ActionID, - 2) missing colon - - * asterisk.c: Bug 6849 - trivial typo fix - -2006-03-30 21:44 +0000 [r16534-16559] Joshua Colp <joshnet@nbnet.nb.ca> - - * codecs/gsm/Makefile: Add another check for 64-bit goodness (issue - #6850 reported by evilbunny) - - * res/res_musiconhold.c: Do not exceed the array size for maximum - allowed moh files. (issue #6842) - -2006-03-30 01:34 +0000 [r16303-16346] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Set initial value on adsipark - - * apps/app_groupcount.c: Typo fix. - - * configs/extensions.conf.sample: Typo (Issue 6839, casper) - -2006-03-29 19:11 +0000 [r16082-16192] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/pbx.h, apps/app_stack.c, pbx.c: Bug 6830 - Let - GosubIf work with the same conditions as a GotoIf (change in API - approved by Russell) - - * pbx.c: Bug 6835 - Updates to GotoIf help text - -2006-03-29 04:15 +0000 [r16008] Russell Bryant <russell@digium.com> - - * strcompat.c: tell unsetenv for solaris to return the result of - the setenv call - -2006-03-29 00:58 +0000 [r15898] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6823 - Portability issue with the - registration port number patch from yesterday. Be compatible with - more systems than OS/X :-) Thanks Rizzo for the advice. - -2006-03-29 00:32 +0000 [r15896] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/linkedlists.h: ensure that list traversal loops - which skip entries properly update the 'previous entry' pointer - so when entries _are_ removed the list does not get damaged - -2006-03-28 20:22 +0000 [r15703-15743] Russell Bryant <russell@digium.com> - - * agi/Makefile, strcompat.c, astmm.c: backport astmm + sparc fixes - from the trunk - - * channels/chan_iax2.c: fix Bus Error on sparc (issue #6354) - -2006-03-28 19:07 +0000 [r15699] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Fix breakage of NAT support for peers with - qualify=yes. Thanks Damin for access to your system, sorry folks. - -2006-03-28 18:09 +0000 [r15658] Russell Bryant <russell@digium.com> - - * pbx/pbx_ael.c: fix the order in which for loops are expanded - (issue #6810) - -2006-03-28 17:48 +0000 [r15615] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/init.d/rc.redhat.asterisk: Bug 6815 - Adding quotes to - make bash happy - -2006-03-27 23:45 +0000 [r15366-15381] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6736 - Use flags for OPTION messages. - Thanks Casper! - - * channels/chan_sip.c: Issue #6597 - sip show registry shows - incorrect port - - * channels/chan_sip.c: Issue #6409 - Use "s" extension when there's - no username in the URI - -2006-03-26 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.6 released - -2006-03-25 05:07 +0000 [r14821-14868] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/init.d/rc.redhat.asterisk: Bug 6601 - More configuration - abilities for the RH init script - - * apps/app_voicemail.c: Fix incorrect size of zeroing (left over - from when maxmsg was hardcoded at 100) - - * apps/app_voicemail.c: Bug 6783 - When context is specified, - voicemail should look for mailboxes in that context - -2006-03-24 14:48 +0000 [r14704] Russell Bryant <russell@digium.com> - - * image.c: use the correct variable in an error message (issue - #6791) - -2006-03-24 04:53 +0000 [r14610-14659] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c: Fix a typo in the app description - - * include/asterisk/sched.h: Doxygen comment typo corrections - -2006-03-23 21:51 +0000 [r14523] Joshua Colp <joshnet@nbnet.nb.ca> - - * res/res_features.c: Issue #6764 - Return BUSY signal when other - party is busy at Attended Transfer (Reported by mnachev) - -2006-03-23 21:44 +0000 [r14522] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Fix SETUP_ACK handling so that we change - channels if so requested - -2006-03-23 20:43 +0000 [r14467] BJ Weschke <bweschke@btwtech.com> - - * apps/app_meetme.c: Bug #5884 - fix a possible race state in - app_meetme when a channel has gone away and we are reading - continuously for more frames. (mneuhauser) - -2006-03-23 20:13 +0000 [r14462] Russell Bryant <russell@digium.com> - - * apps/app_readfile.c: don't crash when asked to read from a file - that doesn't exist (issue #6786) - -2006-03-22 22:18 +0000 [r14191-14276] Joshua Colp <joshnet@nbnet.nb.ca> - - * apps/app_voicemail.c: Fix a minor code issue - - * apps/app_voicemail.c: Issue #6781 - Verbose levels not enforced - in app_voicemail (Reported by flobi) - - * include/asterisk/cdr.h, cdr.c: Issue #5918 - Disposition showing - FAILED even though call is answered successfully (Reported by - tracinet) - - * pbx.c: Issue #6780 - ast_pbx_outgoing_cdr_failed description fix. - (Reported and fixed by casper) - -2006-03-22 09:10 +0000 [r14140] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6766 - fix ;user=phone functionality. - (Reported by alein, fix by russell - thanks!) - -2006-03-21 18:59 +0000 [r13814-13964] Russell Bryant <russell@digium.com> - - * configs/features.conf.sample: add a note explaining how to set - the DYNAMIC_FEATURES variable to allow the use of custom features - (issue #6747) - - * res/res_features.c: fix crash when using the ParkAndAnnounce - application. When using this application, there will be no peer - channel to play the parking announcement to. (issue #6756) - - * funcs/func_strings.c: fix REGEX on strings that contain quotes - (issue #6678) - - * sounds.txt: fix spelling of whiskey - - * apps/app_meetme.c: don't add conference participant if the user - hangs up while recording their name (issue #6661) - - * sample.call: re-add the Account parameter to the sample call file - since it's not really deprecated since the CDR function is no - longer built in - -2006-03-21 06:24 +0000 [r13707-13748] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6714 - Workaround to avoid retrieving - incomplete voicemail message - - * editline/term.c: Do away with some warnings and fix some - indentation - -2006-03-20 17:36 +0000 [r13634] Olle Johansson <oej@edvina.net> - - * channels/chan_iax2.c: Do not overwrite ANI if it's set by IE - (sendani=yes in the peer) - -2006-03-19 09:59 +0000 [r13550] Russell Bryant <russell@digium.com> - - * apps/app_dial.c: revert the change made in revision 12927 in - favor of keeping the original behavior of the option. The - documentation has now been updated to reflect the actual - behavior. (issue #6523) - -2006-03-19 09:25 +0000 [r13547] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Reset global_rtautoclear at sip reload - -2006-03-16 20:05 +0000 [r13279] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * ast_expr2.y, ast_expr2.c: Bug 6737 - Fix compile warning on OS X - -2006-03-16 17:58 +0000 [r13239] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Issue #6690 - clarify progressinband - default setting - -2006-03-16 17:42 +0000 [r13237] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: always use the callerid signalling method - set in the zt_pvt strucutre as opposed to the last one read from - the config file (issue #6734, with mods) - -2006-03-16 06:56 +0000 [r13197] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: To quote giant developers: "Oops". Thanks, - Tony! - -2006-03-15 22:16 +0000 [r13161] Russell Bryant <russell@digium.com> - - * cdr.c: - remove some calculations that will always result in 0 - - if a CDR was never started, don't try to calculate a duration and - consider it failed - -2006-03-15 13:01 +0000 [r13026] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6728: Remove parameters to Event: - header on SUBSCRIBE requests - -2006-03-14 18:41 +0000 [r12925-12927] Russell Bryant <russell@digium.com> - - * apps/app_dial.c: when using the G() option to Dial, fix sending - the called channel to 1 priority beyond what was specified (issue - #6523) - - * apps/app_queue.c: fix a problem with not loading realtime queue - members by always reloading a realtime queue from the database - even if it is found in the list (issue #6680) - -2006-03-12 19:26 +0000 [r12646] Russell Bryant <russell@digium.com> - - * pbx.c: add locking to protect the list of global dialplan - variables - -2006-03-12 17:57 +0000 [r12577] Russell Bryant <russell@digium.com> - - * codecs/gsm/Makefile: fix build on parisc (issue #6704) - -2006-03-10 12:13 +0000 [r12477-12495] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #5937 - Make sure SIP CANCEL's are - re-transmitted - - * channels/chan_sip.c: Issue #6576 - SIP_CODEC not used for early - media (reported by gpapadop73) - -2006-03-08 10:51 +0000 [r12458] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6657 - Ignore 183 session progress - without SDP - -2006-03-07 00:05 +0000 [r12161-12195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 6020 - Race condition where packet could - be lost if first packet on list is acked - - * editline/np/vis.c, editline/readline.c: Bug 6664 - More fixes for - Solaris - -2006-03-06 14:23 +0000 [r12036-12072] Olle Johansson <oej@edvina.net> - - * channel.c: Revert earlier change - - * channel.c: Fix for astmm compilation - -2006-03-06 02:32 +0000 [r11946] Russell Bryant <russell@digium.com> - - * configs/zapata.conf.sample: fix a typo in the description of the - ringtimeout option - -2006-03-05 12:40 +0000 [r11849] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Clear page2 flags at reload too - -2006-03-04 11:45 +0000 [r11778] BJ Weschke <bweschke@btwtech.com> - - * apps/app_mixmonitor.c: Substitute variables in the post_process - string (if it exists) before those variables could possibly - disappear (channel hangup) #6462 - -2006-03-03 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.5 released - -2006-03-03 00:38 +0000 [r11607-11635] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * Makefile: Bug 6638 - Use POSIX command for Solaris - - * build_tools/make_build_h: Bug 6638 - Change from a historic BSD - command to a POSIX command for determining username - - * asterisk.c: Bug 6637 - Fixes for Solaris - - * Makefile: If debugging, the frame pointer is helpful - -2006-03-02 19:05 +0000 [r11528-11561] Russell Bryant <russell@digium.com> - - * res/res_monitor.c: fix inaccurate ack message to ChangeMonitor - action (issue #6630) - - * asterisk.sgml: make the terminology used in the synopsis match - the option description - - * asterisk.sgml: add the -L option to the synopsis on the man page - -2006-03-01 17:41 +0000 [r11479-11503] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_manager.c, cdr/cdr_tds.c, res/res_config_odbc.c, - include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, cdr.c: - Bug 6615 - Fix 64bit conversion errors by using a long int - - * build_tools/make_svn_branch_name: Bug 6618 - Solaris - compatibility fix - -2006-02-28 19:46 +0000 [r11382-11410] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix the output that indicates whether - qualify smoothing is on or not (issue #6608) - - * asterisk.c: adjust the keys directory when astvarlibdir is - specified in asterisk.conf (issue #6602) - - * res/res_agi.c: add a missing newline in the agi app description - (thanks wunderkin!) - -2006-02-27 15:20 +0000 [r11250-11281] Russell Bryant <russell@digium.com> - - * cli.c: don't try to print the help text for a CLI command when - RESULT_SHOWUSAGE is returned if there is no help text available - (issue #6604) - - * channels/chan_sip.c: fix finding realtime peers that are not - dynamic by ip address (issue #6093) - - * channel.c: don't hang up the channel if its state is set to UP - before we return from ast_call (issue #6569) - -2006-02-26 16:26 +0000 [r11165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/logger.h, logger.c: Bug 5950 - reenable queue - log rotation; also, eliminate redundant code - -2006-02-25 19:54 +0000 [r11120] Matt Frederickson <creslin@digium.com> - - * translate.c: Backport of fix to translation optimizations. Thanks - again file! - -2006-02-25 05:08 +0000 [r11058-11089] Kevin P. Fleming <kpfleming@digium.com> - - * translate.c: factor the number of translation steps required into - translation path decisions, so that equal cost paths that require - fewer translations are preferred - - * translate.c: reformat code to fit guidelines remember which - translation paths are multi-step paths - - * channel.c: ensure that spy frame queueing is able to deal with - translation failing for any reason (issue #6546) - -2006-02-23 23:06 +0000 [r10952] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * Makefile: set PWD properly - -2006-02-23 14:57 +0000 [r10736-10863] Kevin P. Fleming <kpfleming@digium.com> - - * dnsmgr.c, include/asterisk/linkedlists.h: backport list handling - fix from trunk (solves memory leak problem in cdr variables and - device state watchers) remove unused variable to silence - compiler warning - - * configs/iax.conf.sample: add comment warning people about trying - to use hostnames/IPs in the sample config - -2006-02-20 23:01 +0000 [r10577] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * app.c: Would be nice to tell people to look in the right file to - increase a constant - -2006-02-20 06:17 +0000 [r10511-10535] Mark Spencer <markster@digium.com> - - * channels/chan_sip.c: Handle ACKing properly (remove gratuitous - -1) - - * channels/chan_iax2.c: Fix numerous places in jitter buffer where - freed memory is referenced - -2006-02-19 18:29 +0000 [r10462-10487] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * formats/format_sln.c: Okay, fseek doesn't return an offset - - * apps/app_voicemail.c: Fix possible lack of initialization of - useadsi - - * formats/format_sln.c: Bug 6539 - Division by two negates error - flag - -2006-02-18 00:17 +0000 [r10409] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * app.c: Bug 6529 - memory leak in ast_play_and_prepend - -2006-02-17 01:55 +0000 [r10301-10368] Russell Bryant <russell@digium.com> - - * jitterbuf.c: fix incorrent index calculation for jitterbuffer - history (issue #6517) - - * apps/app_voicemail.c: when executing the Directory application - from voicemail and a context is not specified, use the "default" - context, not the channel's current context (issue #6507) - -2006-02-15 01:21 +0000 [r10108-10137] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_agent.c: ensure that agents logged in via the - manager interface are stored in the persistence database (related - to issue #6301) - - * funcs/func_enum.c: handle longer ENUM lookup results (issue - #6476) - - * res/res_agi.c: ensure that FastAGI launcher can handle system - call interruption (issue #6449) - -2006-02-14 20:56 +0000 [r10021] Matt O'Gorman <mogorman@digium.com> - - * apps/app_meetme.c: bug fix from 6485 with musiconhold not being - turned off by app_meetme - -2006-02-14 20:20 +0000 [r10018] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: don't double-increment abandon counter for - calls that are hung up while dialing members (issue #6289) - -2006-02-14 19:11 +0000 [r9990] Mark Spencer <markster@digium.com> - - * apps/app_meetme.c: Fix stopstream in menus (bug #6137) - -2006-02-14 18:50 +0000 [r9961-9964] BJ Weschke <bweschke@btwtech.com> - - * asterisk.c: #ifdef the include too. - - * asterisk.c: #ifdef'd the prctl fix to only try and compile on - linux systems. Thanks rizzo for pointing this out. - -2006-02-14 18:30 +0000 [r9953-9958] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: when answering INVITE, don't send codecs the - peer didn't offer (issue #6052) - - * rtp.c: revert yesterday's temporary fix for issue #6052 - -2006-02-14 04:45 +0000 [r9861-9870] BJ Weschke <bweschke@btwtech.com> - - * asterisk.c: Fixed my silly backport error from r9861 - - * asterisk.c: Merged changes from r9844 from /trunk. Make sure that - PR_SET_DUMPABLE is set to make certain that we still dump core if - Asterisk has setuid'd to run as non-root. - -2006-02-14 00:46 +0000 [r9818] Kevin P. Fleming <kpfleming@digium.com> - - * rtp.c: don't try to use peer's dynamic codec numbers, it leads to - duplication (issue #6052) - -2006-02-13 17:37 +0000 [r9756] Josh Roberson <josh@asteriasgi.com> - - * apps/app_meetme.c: Don't set the formats before we stop - indications. (issue #6380) - -2006-02-11 19:23 +0000 [r9581-9609] Russell Bryant <russell@digium.com> - - * channels/chan_mgcp.c, channels/chan_sip.c, pbx/pbx_dundi.c, - channels/chan_iax2.c: fix memory leak from not destroying the - scheduler context on module unload - - * apps/app_page.c: fix due to CDR changes - - * manager.c, pbx/pbx_spool.c, include/asterisk/channel.h, - include/asterisk/pbx.h, include/asterisk/manager.h, channel.c, - pbx.c: now that CDR is a loadable module, don't depend on it - elsewhere (issue #6460) - -2006-02-11 15:22 +0000 [r9528] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, cdr.c: clean up my mess from thread-starting - change - -2006-02-11 06:29 +0000 [r9493] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: kpfleming's fix from r9472 backported to 1.2 - -2006-02-10 20:38 +0000 [r9404] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_mgcp.c, dnsmgr.c, channels/chan_sip.c, - devicestate.c, channels/chan_modem.c, cdr.c: don't create monitor - threads in detached mode, when we need to be able to - pthread_join() them later if the module is unloaded (solve - crash-on-unload problem for these channel modules) - -2006-02-09 21:10 +0000 [r9323-9326] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Revert behavior change from previous commit - (fixes only) - - * apps/app_voicemail.c: Backport 5929 to 1.2 - -2006-02-09 02:31 +0000 [r9246-9262] Russell Bryant <russell@digium.com> - - * apps/Makefile: add another location for postgresql headers (issue - #6419) - - * channels/chan_iax2.c: reload peercontext on iax2 reload (issue - #6442) - -2006-02-08 22:34 +0000 [r9233] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/Makefile: Leave it to RH/CentOS to put the freetds headers in - a completely nonstandard location. - -2006-02-08 22:12 +0000 [r9232] Matt O'Gorman <mogorman@digium.com> - - * logger.c, channels/chan_oss.c: Make logger report - error,warning,notice if logger.conf not found, also updated - chan_oss to give correct error message if its config file is not - found. - -2006-02-05 17:10 +0000 [r9156] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_macro.c: Bug 6176 - Fix race condition - -2006-02-02 18:37 +0000 [r9086] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't override ASTERISKVERSIONNUM to 000000 for non-svn - builds - -2006-02-02 16:12 +0000 [r9073] Matt Frederickson <creslin@digium.com> - - * res/res_odbc.c: Fix for (#6309), potential (highly unlikely) - memory leak in res_odbc - -2006-01-30 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.4 Released - -2006-01-30 17:08 +0000 [r8905] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: disable buggy PRI user-user code until it - can be fixed - -2006-01-28 13:52 +0000 [r8808] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6182 - Don't remove scheduled event - until it's really done. (reported by malverian) - -2006-01-27 08:02 +0000 [r8785] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6362 - Register without Contact: and - Expires: fails (reporter: op) - -2006-01-27 00:52 +0000 [r8758] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * ast_expr2.h, ast_expr2f.c, ast_expr2.c: Bug 6072 - Revisions to - the source bison and flex files don't auto-regenerate these files - -2006-01-26 19:42 +0000 [r8729] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: fix problem with dtmf on e&m (issue #6364) - -2006-01-26 14:39 +0000 [r8710] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 5898: Registrations does not get - deleted if there's an active SIP dialog - -2006-01-25 19:14 +0000 [r8666-8677] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: don't call ast_update_realtime with - uninitialized variables if we get a registration with an expirey - of 0 seconds (issue #6173) - - * channels/chan_features.c: fix memory leak (inspired by issue - #6351) - -2006-01-25 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.3 Released - -2006-01-25 09:46 +0000 [r8632] Olle Johansson <oej@edvina.net> - - * channel.c: Issue #6439 - the "timebomb" bug. Patch by Markster - over GPRS - -2006-01-25 05:38 +0000 [r8619] Russell Bryant <russell@digium.com> - - * utils/astman.c: don't leak almost 200 bytes for each new channel - (issue #6330) - -2006-01-25 01:50 +0000 [r8608] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c: ensure hangup cause code is handled properly - when channel does not return a frame (issue #6346) - -2006-01-24 22:55 +0000 [r8600] Russell Bryant <russell@digium.com> - - * asterisk.c: completely arbitrary whitespace change for testing - something with svnmerge ... - -2006-01-24 22:32 +0000 [r8588] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: ensure that channel cannot become zombie after we - check but before we try to start indications - -2006-01-24 20:37 +0000 [r8573] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Backport fix for #6229, hangup on polarity - reversal - -2006-01-24 19:21 +0000 [r8537-8562] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6114: Don't hangup on BYE/ALSO with no - channel. - - * channels/chan_sip.c: Issue #6308 - never send response to ACK. - (Reported by whiskerp) - -2006-01-22 19:03 +0000 [r8437-8445] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: fix memory leak from not freeing the queue - member list when freeing an old queue - - * channel.c: fix MixMonitor crash (issue #6321, probably others) - -2006-01-22 15:13 +0000 [r8433] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: Bug fix: Correct some scenarios where - CALL_LIMIT could not be getting adjusted properly allowing - chan_sip to send calls when it really shouldn't. Bug #6111 - -2006-01-22 08:52 +0000 [r8429] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 6281 - Cannot set more than a single - header with SIPAddHeader - -2006-01-22 02:05 +0000 [r8412-8418] Russell Bryant <russell@digium.com> - - * pbx.c: add a modified fix to prevent writing outside of the - provided workspace when calculating a substring (issue #6271) - - * pbx.c: temporarily revert substring fix pending the result of the - discussion in issue #6271 - - * pbx.c: prevent the possibility of writing outside of the - available workspace (issue #6271) - -2006-01-21 18:29 +0000 [r8394] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_queue.c: Bug 5936 - AddQueueMember fails on realtime - queue, if queue not yet loaded - -2006-01-20 18:34 +0000 [r8347] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: fix invalid value of prev_q (issue #6302) - -2006-01-20 01:00 +0000 [r8320] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_iax2.c: solved problem with delayreject and iax - trunking bug 4291 - -2006-01-19 19:40 +0000 [r8281] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Enable "musicclass" setting for sip peers as - per the config sample. - -2006-01-19 19:14 +0000 [r8276] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * ast_expr2.y, ast_expr2.fl: Bug 6072 - Memory leaks in the - expression parser - -2006-01-19 04:56 +0000 [r8232-8242] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix Message-Account header to use the ip - address if the fromdomain isn't set (issue #6278) - - * apps/app_milliwatt.c: fix a seg fault due to assuming that space - gets allocatted on the stack in the same order that we declare - the variables (issue #6290) - -2006-01-18 21:02 +0000 [r8194] Matt O'Gorman <mogorman@digium.com> - - * apps/app_meetme.c: Solves issue with the login proccess in meetme - patch from 6136 - -2006-01-18 02:49 +0000 [r8173] Russell Bryant <russell@digium.com> - - * ChangeLog (removed): remove ChangeLog from the 1.2 branch. It - will only be present in the tags. - -2006-01-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.2 Released - -2006-01-18 00:47 +0000 [r8140-8162] Matt O'Gorman <mogorman@digium.com> - - * loader.c: Changed order of autoload so that pbx_ comes before - channels, and in doing so cause bug 6002 to not be an issue - - * apps/app_festival.c: Stop any generators running on a channel - when festival is called as described in 5996 - -2006-01-17 18:29 +0000 [r8134] Matt Frederickson <creslin@digium.com> - - * res/res_features.c: Backport of fix for #6094 - -2006-01-17 16:55 +0000 [r8124] Matt O'Gorman <mogorman@digium.com> - - * logger.c: Fixed code ordering of logger_init and queue_log_init - bug 6263 - -2006-01-17 13:11 +0000 [r8112-8122] Kevin P. Fleming <kpfleming@digium.com> - - * asterisk.c: update CLI copyright notice - - * asterisk.c: do rlimit check _after_ reading config file, in case - 'dumpcore' is specified there - -2006-01-14 19:06 +0000 [r8074] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Bug 6238 - Fix segfault when delimiter not - specified - -2006-01-13 06:07 +0000 [r8047] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: fix spelling errors (issue #6227) - -2006-01-12 06:14 +0000 [r7999] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, configs/voicemail.conf.sample: Bug 6211 - - Add option deletevoicemail as equivalent to option delete for - Realtime - -2006-01-11 19:08 +0000 [r7965-7986] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: move variable to correct scope (issue - #6197) - - * apps/app_voicemail.c: fix temp greetings with ODBC storage (issue - #6078) - - * channels/chan_sip.c: fix mem leak on module unload (issue #6190) - - * app.c: don't override an error condition that occurred when - acting on the primary channel when stopping the autoservice on - the peer channel. (from issue #6087) - - * translate.c: lock list of translators *before* recalculating the - translation matrix - -2006-01-11 04:38 +0000 [r7963] Matt O'Gorman <mogorman@digium.com> - - * channel.c: Minor typo refrenced in 6191 - -2006-01-11 04:19 +0000 [r7957-7960] Russell Bryant <russell@digium.com> - - * pbx.c: fix locking error - lock instead of unlock - - * apps/app_dial.c: fix a little typo - -2006-01-11 01:30 +0000 [r7955] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6192 - behave correctly when mailbox is - specified as argument - -2006-01-10 08:48 +0000 [r7939] Olle Johansson <oej@edvina.net> - - * doc/README.cdr: - Adding reference to README.tds - Reformatting - table - -2006-01-09 22:48 +0000 [r7917] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: re-initialize _all_ sequence numbers when - transfer completes - -2006-01-09 22:07 +0000 [r7915] Russell Bryant <russell@digium.com> - - * file.c: add missing unlock (issue #6112) - -2006-01-09 20:08 +0000 [r7904-7908] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_spool.c: Bug 6157 - Memory leak - - * doc/README.variables: Update variable documentation to match the - code - -2006-01-09 18:11 +0000 [r7898-7900] Kevin P. Fleming <kpfleming@digium.com> - - * asterisk.c: commit user/group-related changes from trunk - - * db.c: backport fix from revision 7856 of trunk - - * apps/app_voicemail.c: fix breakage introduced in revision 7871 - -2006-01-09 05:11 +0000 [r7870-7871] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: fix seg fault when using greek syntax in - VoicemMailMain (issue #6142) - - * manager.c: backport fix for unnecessary unlock (issue #6171) - -2006-01-07 07:27 +0000 [r7848] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_spool.c: Bug 6156 - catch all threading errors, not just - simple failure - -2006-01-06 00:34 +0000 [r7831] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_config.c: Dumb error messages - "Context 'context' - already included in 'in' context" - -2006-01-06 00:21 +0000 [r7829] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_agent.c: update agent persistence when an agent - gets logged off by autologoff - -2006-01-05 23:53 +0000 [r7827] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/strings.h: Bug 6076 - Fix documentation of - ast_trim_blank return value - -2006-01-05 23:49 +0000 [r7825] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: eliminate rounding errors that caused call time limits - to be inaccurate (issue #5913) round 'time left' reported during - call limit warnings up to sound more accurate - -2006-01-05 23:07 +0000 [r7823] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_features.c: Bug 6081 - fix for memory leak, formatting - fixes - -2006-01-05 20:52 +0000 [r7819] Kevin P. Fleming <kpfleming@digium.com> - - * formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that - variable is initialized - -2006-01-05 09:13 +0000 [r7812] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Fix copyright of changed file - -2006-01-05 00:58 +0000 [r7799-7809] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_agent.c: send device state updates for auto-logoff - of agents as well - - * formats/format_pcm.c, formats/format_pcm_alaw.c: doh... fseek() - has no useful return value - - * formats/format_pcm.c, formats/format_pcm_alaw.c: use proper - fwrite() parameters and return value - - * formats/format_pcm.c, formats/format_pcm_alaw.c: return properly - after extending file - - * formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that - ulaw/alaw sound files are filled with silence when extended (not - zeroes) - - * channel.c: make monitoring more tolerant of peers that deliver - frames in bursts - -2006-01-04 21:46 +0000 [r7792-7795] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Issue #5980: Removing extra CR+LF in manager - events - needs port to trunk - - * channels/chan_sip.c: Fixing typo in XML for video updates. - -2006-01-04 07:06 +0000 [r7773] Russell Bryant <russell@digium.com> - - * funcs/func_moh.c: use a more correct way of determining the size - of the destination buffer - -2006-01-04 05:27 +0000 [r7771] BJ Weschke <bweschke@btwtech.com> - - * apps/app_privacy.c: Fix the 'if' clause to be true under the - right conditions. Bug #6126 - -2006-01-03 20:22 +0000 [r7746] Kevin P. Fleming <kpfleming@digium.com> - - * ast_expr.y (removed): remove unused 'old' expression parser - -2006-01-03 18:15 +0000 [r7743] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_stack.c: Bug 6121 - typo in application description - -2006-01-03 17:24 +0000 [r7736-7740] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/chanspy.h, apps/app_chanspy.c, - apps/app_mixmonitor.c, channel.c: revert incorrect fix for bug - #6048 from revision 7709 put in correct (simpler) fix add doxygen - docs for channel spy 'state' values - - * channels/chan_sip.c: backport rport scanning fix from trunk (bug - #6071) - - * ast_expr2f.c, ast_expr2.fl: don't leak memory for (most) - expression evaluations - -2006-01-02 07:31 +0000 [r7709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_mixmonitor.c: Bug 6084 - MixMonitor after a 'cli stop - monitor' deadlocks - -2006-01-02 02:04 +0000 [r7706] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Fix compiler warnings. - -2005-12-30 14:54 +0000 [r7677] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channel.c: Bug 6091 - Fix race condition around uniqueid - -2005-12-28 17:35 +0000 [r7663-7665] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix memory leak in build_rpid (issue #6070) - - * apps/app_chanspy.c: backport fix for permissions of created - recordings (issue #6067) - -2005-12-27 00:07 +0000 [r7641] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: backport fix to ensure that DSP is never - enabled on pseudo channels - -2005-12-26 20:32 +0000 [r7637] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_tds.c: Remove copy of code in libc, preferring code in - utils.c (public domain code) - -2005-12-26 18:19 +0000 [r7634] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, channels/chan_agent.c, apps/app_sms.c, - asterisk.c, config.c, pbx/pbx_dundi.c, apps/app_externalivr.c, - apps/app_queue.c, channels/chan_iax2.c, cli.c, - apps/app_chanspy.c, res/res_monitor.c: cast time_t to an int in - printf/scanf (issue #5635) - -2005-12-23 06:38 +0000 [r7608] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_hasnewvoicemail.c: Bug 6051 - VMCOUNT should work as - documented and count all, not quit after finding 1 - -2005-12-23 03:01 +0000 [r7606] Kevin P. Fleming <kpfleming@digium.com> - - * asterisk.c: add license reference to copyright notice displayed - when CLI session begins add 'show warranty' and 'show license' - CLI commands (still need a complete list of non-GPL components - included in Asterisk) - -2005-12-23 00:00 +0000 [r7605] BJ Weschke <bweschke@btwtech.com> - - * apps/app_waitforsilence.c: Another app documentation tweak. - -2005-12-22 22:04 +0000 [r7601] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6050 SQL requires the use of single - ticks to delimit values, not quotes - -2005-12-22 20:36 +0000 [r7595-7599] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: revert changes to - videosupport to allow per-peer setting, since it isn't quite - complete and there is not an obvious fix at this point - - * channels/chan_sip.c: remove stray unlock (issue #5955) - -2005-12-21 22:23 +0000 [r7586] Josh Roberson <josh@asteriasgi.com> - - * channels/chan_sip.c: Actually put in the per-peer settings for - sip video, as they didn't make it in at astricon somehow, and - I've been too busy up until now to redo it. - -2005-12-21 20:01 +0000 [r7582] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_alsa.c: Allow a chan_alsa that failed to open sound - devices to be unloaded. - -2005-12-21 19:53 +0000 [r7580] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Bug #6040 - Documentation correction - -2005-12-21 19:23 +0000 [r7577] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_ael.c: Bug 5777 - Remove parentheses on Goto in AEL, so - that it parses correctly - -2005-12-20 20:21 +0000 [r7550-7557] Russell Bryant <russell@digium.com> - - * res/res_agi.c: check array bounds when parsing arguments to AGI - (issue #5868) - - * channels/chan_iax2.c: backport fix for reloading peer context - (issue #6007) - - * apps/app_directed_pickup.c: backport fix for segfault on directed - pickup when no CDR is available (issue #5998) - -2005-12-20 12:58 +0000 [r7546] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_meetme.c: backport fix for larger-than-20ms-frames from - trunk (bug #5697) - -2005-12-19 23:47 +0000 [r7529] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: I messed up and accidently committed this to - the trunk first ... - add note on required values of sip_methods - struct - remove duplicate function prototype - remove duplicate - ast_mutex_lock (issue #6025) - -2005-12-19 19:06 +0000 [r7521-7523] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * file.c: Bug 5988 - record append option not working - - * cdr.c: Bug 6026 - segfault for the sequence NoCDR(), - SetAMAFlags() - -2005-12-17 18:55 +0000 [r7517-7519] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * doc/README.ael: Document that curley braces must be on the same - line as the keyword. - - * apps/app_chanspy.c: Bug 6009 - off by one error - -2005-12-17 03:59 +0000 [r7510-7515] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: Max-Forwards headers must only be present on - requests, not responses - - * channels/chan_sip.c: forcibly expire previous subscriptions from - a peer when they resubscribe (keeps them from building up and - waiting for expiration, and stops us sending unwanted NOTIFY - messages to devices) - - * build_tools/make_svn_branch_name: fix some buglet when building - team branch version strings - -2005-12-17 01:02 +0000 [r7508] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: We want to check the previous - value, not the current value (which was just changed). - -2005-12-16 00:49 +0000 [r7497] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_cut.c: First field is truncated - -2005-12-15 10:52 +0000 [r7490] Christian Richter <christian.richter@beronet.com> - - * doc/README.misdn, channels/misdn/mISDNuser.patch (added), - channels/misdn/isdn_lib_intern.h, channels/misdn/mISDN.patch - (added), channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/Makefile, channels/misdn/chan_misdn_config.h, - channels/misdn/ie.c, channels/chan_misdn_config.c, - channels/misdn/isdn_msg_parser.c, channels/Makefile, - channels/misdn/isdn_lib.c: * Added mISDN/mISDNuser Echo cancel - Patch * Fixed Makefiles so that chan_misdn can be compiled again - * added some hints, that mISDN cannot be compiled against gcc-4, - SMP, Spinlock Debug * fixed some Minor issues in chan_misdn, - regarding Type Of Number and Presentation - -2005-12-15 02:51 +0000 [r7482] BJ Weschke <bweschke@btwtech.com> - - * channel.c: Bug #6003 - Don't free the channel structure until - after having sent the manager event. - -2005-12-13 18:54 +0000 [r7435-7470] Kevin P. Fleming <kpfleming@digium.com> - - * doc/README.variables: clarify substring documentation - - * utils.c: correct broken math in tvfix() for timestamp values over - one million - - * apps/app_dial.c: restore ability of caller to hangup calls that - are still ringing (issue #5839) - - * channels/chan_sip.c, pbx.c: ensure that hangups while incoming - calls are in early state are handled properly (issue #5919) - - * channels/chan_agent.c: only report AGENT_IDLE for callback mode - agents when they are actually idle (issue #5902) - - * app.c: use the stream's current point when pausing/unpausing, - instead of elapsed time (which doesn't work when the stream has - been skipped forward or backward) (issue #5897) - - * apps/app_externalivr.c: set all the child file descriptors to - non-blocking so that we don't hang if the child fails to send a - newline-terminated command or error message - -2005-12-12 17:19 +0000 [r7433] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: Typo - -2005-12-11 06:08 +0000 [r7430] Russell Bryant <russell@digium.com> - - * utils/astman.c: silence a couple of compiler warnings about - pointer signedness - -2005-12-11 01:26 +0000 [r7427-7429] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: Bug 5965 - major bug in - AST_LIST_REMOVE - - * apps/app_voicemail.c: Bug 5967 - -2005-12-10 18:10 +0000 [r7425] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_zap.c: Bug #5877 Make sure the digit string from - E&M wink DNIS collection is properly null terminated as it grows. - -2005-12-08 23:45 +0000 [r7404-7406] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 5960 - - * configs/res_odbc.conf.sample: Documenting two keywords that were - previously missing - -2005-12-08 01:05 +0000 [r7382-7386] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: initialize the buffer before using it... - - * pbx.c: ensure that hints are allowed to use global variable - references - -2005-12-06 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.1 Released - -2005-12-05 06:47 +0000 [r7335-7340] Russell Bryant <russell@digium.com> - - * Makefile: remove ASTERISKVERSIONNUM from the version string given - to doxygen - - * apps/app_queue.c: don't delete dynamic queue members when - reloading the static members from a realtime database (issue - #5922) - - * channels/chan_sip.c: fix the order of arguments to an error - message (issue #5927) - -2005-12-04 18:03 +0000 [r7329] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_svn_branch_name: use a more efficient way to get - the revision number, that will also report if the working copy - contains uncommitted modifications - -2005-12-03 19:55 +0000 [r7310] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 5925: check for "Unknown", as that's - what app_voicemail puts into the field for Unknown callerid Also, - remove useless res checks (initialized to 0; never set) - -2005-12-03 01:24 +0000 [r7299] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Documenting the default registerattempts - setting as 0, continue hammering the server for ever and ever ;-) - -2005-12-02 21:12 +0000 [r7285] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * 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: Turn on executable bits for - startup scripts, and fix bash var interpolation for Mandrake - -2005-12-02 00:52 +0000 [r7275] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Bug #5907. Improve SIP INFO DTMF debugging - output. (1.2 & Trunk) - -2005-12-02 00:51 +0000 [r7266-7274] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_page.c, pbx.c: inherit channel variables into channels - created by Page() application (issue #5888) - - * apps/app_voicemail.c, configs/voicemail.conf.sample, UPGRADE.txt: - allow previous context-searching behavior to be used if desired - (issue #5899) - - * apps/app_voicemail.c: properly handle password changes when - mailbox is last line of config file and not followed by a newline - (issue #5870) reformat password changing code to conform to - coding guidelines (issue #5870) - - * channels/chan_agent.c: protect agent_bridgedchannel() from - segfaulting when there is no bridged channel (issue #5879) - - * channels/chan_local.c: allow variables to exist on both 'halves' - of the Local channel (issue #5810) - - * apps/app_festival.c: don't block waiting for the Festival server - forever when it goes away (issue #5882) - - * channel.c: ensure channel's scheduling context is freed (issue - #5788) - - * Makefile, patches (removed): Makefile 'update' target now - supports updating from Subversion repositories (issue #5875) - remove support for 'patches' subdirectory, it's no longer useful - -2005-12-01 23:18 +0000 [r7261-7265] Olle Johansson <oej@edvina.net> - - * doc/README.misdn: Changing bug report address to the Asterisk - issue tracker - - * doc/README.jitterbuffer, doc/README.realtime: Removing references - to 1.1dev, replacing with 1.2, in documentation files. - - * doc/README.misdn: Fixing some spelling errors, as well as - changing "cvs" to "subversion" in misdn documentation. - -2005-12-01 19:25 +0000 [r7257] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_svn_branch_name: ensure that 'svn info' output - is in the expected language for the script to parse (issue #5880) - -2005-12-01 02:33 +0000 [r7228-7251] Russell Bryant <russell@digium.com> - - * apps/app_externalivr.c: use ast_app_separate_args to split - arguments (issue #5686) - - * apps/app_queue.c: fix queue weight feature - compare member - interfaces instead of pointers to the members, since each queue - has its own list of members. (issue #5863) - - * build_tools/make_svn_branch_name: use '=' instead of '==' for - string comparisons. /bin/bash is ok with this, but /bin/sh is - not. (issue #5885) - - * redhat/asterisk (removed), Makefile: remove outdated redhat init - script and provide the updated one in 'make rpm' (issue #5786) - - * contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.redhat.asterisk: Comment out LD_ASSUME_KERNEL - by default. Print error messages if the asterisk executable or - the asterisk configuration directory are not found. (issue #5785, - #5708) - - * apps/app_dial.c: fix DIALEDTIME when call has not been answered - (issue #5862) - - * rtp.c: do not allow an rtp message with zero type (issue #5749) - - * pbx.c: fix hint case sensitivity (issue #5856) - - * configs/sip.conf.sample: add description of the "fromdomain" - option (issue #5874) - -2005-11-30 03:52 +0000 [r7227] Josh Roberson <josh@asteriasgi.com> - - * apps/app_voicemail.c, UPGRADE.txt, ChangeLog: backport fix from - trunk - -2005-11-30 03:37 +0000 [r7219-7226] Kevin P. Fleming <kpfleming@digium.com> - - * doc/cdr.txt, doc/CODING-GUIDELINES, include/asterisk.h, - doc/README.mp3: remove remaining CVS references - - * channel.c: port memory leak fix from rev 7223 in trunk - - * include/asterisk/lock.h: do the multiple-lock check for cond_wait - properly... - -2005-11-29 06:12 +0000 [r7216-7218] Russell Bryant <russell@digium.com> - - * apps/app_cut.c: print an error message if invalid arguments are - specified - - * apps/app_skel.c: fix a couple of typos and a buglet - -2005-11-29 01:25 +0000 [r7199-7213] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/lock.h: if the lock protected a pthread_cond is - held recursively, warn before waiting onthe condition - - * Makefile, build_tools/make_svn_branch_name (added): port version - string computation from trunk - - * / (added): branch renames remove unneeded branches - -2005-11-29 Josh Roberson <josh@asteriasgi.com> - - * apps/app_voicemail.c: Only look in 'default' context when no context defined to VoiceMailMain(). (issue #5887) - -2005-11-25 Russell Bryant <russell@digium.com> - - * apps/app_dial.c: Properly duplicate the string for ANI (issue #5850) - -2005-11-23 Russell Bryant <russell@digium.com> - - * configs/voicemail.conf.sample: Add note to indicate that #include should not be used for this file. (issue #5828) - - * indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - * configs/indications.conf.sample: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - * include/asterisk/indications.h: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - * res/res_indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - - * apps/app_voicemail.c: Remove left over "yay!" debugging message. (issue #5829) - -2005-11-21 Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_cut.c: remove unnecessary include that causes spurious rebuilding - - * channels/chan_sip.c (build_peer): ensure that case changes made to peer names are not ignored during reload operations - (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified - - * channels/chan_iax2.c (build_peer and build_user): ensure that case changes made to peer/user names are not ignored during reload operations - (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified - -2005-11-21 Russell Bryant <russell@digium.com> - - * Makefile: Revert previous change for Darwin. - - * apps/app_osplookup.c: Properly populate the number of results. (issue #5789) - - * Makefile: Don't hard-code that poll functionality needs to be provided on Darwin. - * apps/Makefile: Fix incorrect portion of the patch to fix 'make install' on Solaris. - - * channels/chan_iax2.c (iax2_getpeername): Return non-zero to indicate that a peer was found when using realtime (issue #5815) - -2005-11-20 Russell Bryant <russell@digium.com> - - * Makefile apps/Makefile: Fix 'make install' for Solaris. (issue #5775) - - * apps/app_record.c: Don't leak a frame if writing it to the file fails. (issue #5787) - - * Makefile: Create the monitor spool directory when the other spool directories are created. - - * channels/chan_sip.c channels/chan_iax2.c: Change warning messages about the number of scheduled events happening all at once to debug messages. (issue #5794) - - * pbx/pbx_spool.c: Fix crash when a value is not specified with a variable on a Set: line in a call file. (issue #5806) - - * apps/app_meetme.c: Fix the 'X' option to the MeetMe application. (issue #5773) - - * apps/app_voicemail.c: Correct the use of a mailbox entered by the calling party instead of indicated as an argument to the Voicemail application. (issue #5774) - - * apps/app_controlplayback.c: Fix logic in checking for success when jumping to priority n+101. - * apps/app_md5.c: Fix logic in checking for success when jumping to priority n+101. - - * apps/app_hasnewvoicemail.c: Fix a typo in the application description. Also, fix the logic in checking for success when jumping to priority n+101. (issue #5795) - - * UPGRADE.txt: Add a note on a second way that the IAX2 channel naming convention has changed. (issue #5792) - * channels/chan_iax2.c: Fix alignment of the output for the "iax2 show peer <peer>" CLI command (issue #5792) - - * channels/Makefile: Re-add chan_oss to the default build. (issue #5799) - - * res/res_musiconhold.c: Fix incorrect argument for the buffer size to an ast_copy_string call (issue #5803) - - * funcs/func_enum.c: Shorten the module description (issue #5791) - -2005-11-17 Russell Bryant <russell@digium.com> - - * Makefile: Fix the output of Makefile generated variables to doxygen - - * channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780) - -2005-11-16 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0 released. - -2005-11-16 Jeremy McNamara <jj@nufone.net> - - * apps/app_voicemail.c (load_config): do not terminate asterisk if no voicemail config file - * channels/chan_skinny: Don't register channel type until ready, code formatting updates - -2005-11-16 Josh Roberson <josh@asteriasgi.com> - - * Makefile: Update to fix non-responsive remote console on Darwin (OSX)(issue #5757) - -2005-11-16 Kevin P. Fleming <kpfleming@digium.com> - - * channels/Makefile: don't build chan_modem and sub-modules by default - * configs/modules.conf.sample: explicitly 'noload' chan_modem.so and submodules, in case old versions exist - - * res/Makefile: issue mpg123 not-installed warning at 'make install' time, not 'make' - - * apps/app_forkcdr.c (forkcdr_exec): issue warning (and don't segfault) if ForkCDR is called on a channel that doesn't have a CDR (issue #5763) - - * channel.c (ast_queue_hangup): ensure that the channel lock is held before changing its fields... (issue #5770) - - * res/res_musiconhold.c: don't spit out incorrect log messages (and leak memory) during reload (issue #5766) - - * channels/chan_sip.c (process_sdp): don't pass video codec number into ast_getformatname(), it is not valid input for that function (issue #5764) - - * pbx/pbx_ael.c (match_assignment): properly parse equal signs surrounded by whitespace (issue #5761) - - * doc/README.realtime: document the limitations of using FreeTDS with Realtime (issue #5767) - -2005-11-15 Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: use -g3 for compiler to include macro information for debugger - - * astmm.c (__ast_vasprintf): don't re-use the ap list without copying it; that's not safe on some platforms (issue #5035) - - * doc/README.backtrace: add note about properly building Asterisk to be able to produce backtraces; wrap text and remove DOS line endings - - * channels/chan_sip.c (add_codec_to_sdp): add 'annexb=no' to G.729A SDP (issue #5539) - - * channels/chan_alsa.c (alsa_hangup): handle autohangup properly (issue #5672) - - * channels/chan_misdn.c (and other files): various fixes (issue #5739) - - * channels/chan_sip.c (handle_request_info): properly forward 'flash' events received via SIP INFO (issue #5751, different patch) - - * apps/app_disa.c (disa_exec): don't duplicate constant strings when not needed - - * apps/app_playback.c (playback_exec): use correct logic tests for options (issue #5752) - - * apps/app_disa.c (disa_exec): use standard arg parsing routines (issue #5736) - -2005-11-15 Russell Bryant <russell@digium.com> - - * manager.c: Don't crash on a SetVar action if the channel name is not set, or variable's value is not set (issue #5760) - - * doc/README.variables: Add application exit status variables - -2005-11-14 Josh Roberson <josh@asteriasgi.com> - - * manager.c: Fix crash on variable passing from AMI originate (issue #5737) - -2005-11-14 Russell Bryant <russell@digium.com> - - * many files: Merge doxygen documentation updates. (issue #5605) - - * apps/app_dial.c: Fix typo in RetryDial description. - -2005-11-12 Russell Bryant <russell@digium.com> - - * channels/chan_oss.c: Fix a typo in an error message. - -2005-11-11 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0-rc2 released. - -2005-11-11 Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c (thread_safe_rand): ensure that threads don't get the same random number (issue #5712) - - * apps/app_voicemail.c (forward_message): correct bugs in message forwarding (issue #5718) - (copy_message): use correct path for locking (issue #5704) - - * apps/app_dial.c (wait_for_answer): correct flag copying for automon feature (issue #5720) - - * channels/chan_iax2.c: correct comment - - * apps/app_voicemail.c (close_mailbox): correct previous commit (issue #5663) - (vm_change_password): fix password change writing (issue #5721) - - * channels/chan_sip.c (transmit_invite): remove useless debug message; don't try to add OSP tokens to OPTIONS pings - - * apps/app_voicemail.c (close_mailbox): properly remove deleted messages at mailbox close time (issue #5663) - -2005-11-11 Mark Spencer <markster@digium.com> - - * channels/chan_zap.c (zt_bridge): only enable/disable DTMF detection on SUB_REAL channels - -2005-11-10 Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that system headers that provide basic types are included first (issue #5713) - -2005-11-11 Russell Bryant <russell@digium.com> - - * many files in apps/: Clean up application descriptions. Clarify some wording and make sure they wrap at 80 characters. - -2005-11-10 Mark Spencer <markster@digium.com> - - * rtp.c (ast_rtp_raw_write): use unsigned int for return value from calc_txstamp() (issue #5595) - (calc_txstamp): never return a value that was less than zero before being turned into 'unsigned int' (issue #5595) - -2005-11-10 Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/chanspy.h: move spy-related stuff into separate header so chan_h323 can build (issue #5590) - - * include/asterisk/linkedlists.h (AST_LIST_HEAD_SET_NOLOCK): properly initialize tail pointer when list head is directly set (issue #5669) - - * app.c (ast_app_parse_options): ok, so we aren't all perfect... let's make the while loop actually work properly here (issue #5684) - - * apps/app_disa.c (disa_exec): correct password file parsing (issue #5676) - - * apps/app_meetme.c (conf_run): don't restrict admin users from joining a locked conference (issue #5680) - - * channels/chan_misdn.c: include stdio.h (issue #5671) - * channels/chan_misdn_config.c: fix prototype warning (issue #5671) - - * pbx.c: remove apps that were deprecated before 1.0 was released (issue #5673) - - * apps/app_striplsd.c, apps/app_substring.c: remove apps that were deprecated before 1.0 was released (issue #5673) - - * include/asterisk/lock.h (PTHREAD_MUTEX_RECURSIVE_NP): work around header problems on Cygwin (issue #5668) - - * pbx/pbx_ael.c: handle switch default cases inside macros properly (issue #5354) - - * configs/voicemail.conf.sample (format): add strong warning about changing format list when mailboxes contain messages (issue #5689) - - * many files: ensure that system headers are included before Asterisk headers (issue #5693) - - * channels/chan_iax2.c (complete_iax2_show_peer): don't return from function without releasing lock (issue #5685) - - * channels/iax2-provision.c (iax_provision_reload): don't leak memory (issue #5700) - - * pbx/pbx_ael.c (handle_macro): don't leak memory (issue #5701) - (handle_context): ditto - - * res/res_features.c (load_config): properly initialize referenced variable (issue #5703) - - * apps/app_queue.c (rqm_exec): correct segfault problem (issue #5705) - (aqm_exec): ditto - - * app.c (ast_app_parse_options): don't increment 's' until after checking for NULL (related to issue #5630) - - * apps/app_rpt.c: solve a memory leak (config structure was not freed) (issue #5706) - -2005-11-10 Russell Bryant <russell@digium.com> - - * app.c (ast_app_separate_args): Don't consider the open parenthesis as part of the arguments to an option. (issue #5630) - - * many files: Change all references to ast_separate_app_args to ast_app_separate_args - - * many files in apps/: Clean up some application descriptions. Make sure all descriptions in changed files are wrapped at 80 characters. - -2005-11-09 Russell Bryant <russell@digium.com> - - * pbx.c: Clean up descriptions of built-in dialplan applications. Changes include clearer wording and not referring to return values. - -2005-11-09 Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c (update_registry): don't complain about unspecifed registration expiration intervals, just use the minimum - -2005-11-08 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0-rc1 released. - - * include/asterisk/file.h: add test to ensure that stdio.h is included before this file (issue #5658) - - * res/res_odbc.c (odbc_prepare_and_execute): add new API call for use to properly handle prepared statements across server disconnects (issue #5563) - - * pbx.c (pbx_substitute_variables_helper_full): use already-substituted buffer for parsing variable name (issue #5664) - - * channels/chan_zap.c (zt_request): return AST_CAUSE_CONGESTION when a group-channel is requested and the group exists but all channels are busy (issue #3360, related fix) - * channels/chan_iax2.c (create_addr): treat UNREACHABLE as AST_CAUSE_UNREGISTERED so that it will generate CHANUNAVAIL from app_dial (issue #3360) - - * res/res_features.c (ast_bridge_call_thread_launch): set SCHED_RR separately from thread creation, so it won't fail when running as non-root (issue #5601, different fix) - - * pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720) - * apps/Makefile (APPS): add app_stack (issue #2720) - * apps/app_stack.c: new applications (issue #2720) - - * apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252) - * configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels - - * channels/chan_local.c (local_call): move channel variables from incoming to outgoing instead of inheriting them (issue #5604) - - * many files: add explicit include of stdio.h (issue #5650) - -2005-11-07 Kevin P. Fleming <kpfleming@digium.com> - - * UPGRADE.txt (Parking): add note about new parking behavior (issue #5532) - - * many files: more Cygwin compatibility, and proper getloadavg() prototype/macro (issue #5569) - - * include/asterisk/lock.h (__ast_pthread_mutex_lock): correct build with DETECT_DEADLOCKS defined (issue #5570) - -2005-11-07 Russell Bryant <russell@digium.com> - - * apps/app_queue.c: upgrade to new arg/option API and implement priority jumping control (issue #5580) - * many files: Add missing include of stdio.h, and remove some duplicate and unused header includes - - * include/asterisk/app.h: Increment the arg_index in the options structure to fix applicaiton options that have arguments to them - -2005-11-07 Kevin P. Fleming <kpfleming@digium.com> - - * cryptostub.c: include necessary headers - * include/asterisk/crypto.h: don't include unnecessary headers - - * manager.c (action_setvar): add support for setting global variables (issue #5571) - - * Makefile: correct cross-compilation issue introduced in Cygwin patches (issue #5572) - - * apps/app_voicemail.c: upgrade to new arg/option API and implement priority jumping control (issue #5649) - - * asterisk.c (main): setpriority() failure is not a reason to stop the process (issue #5581) - - * say.c (ast_say_date_with_format_da): say hours properly (issue #5576) - - * manager.c (astman_get_variables): restore old multiple-variable behavior for "Variable" header (issue #5585) - - * many files: don't check for NULL before calling ast_strlen_zero, it can do it itself (issue #5648) - - * pbx.c (handle_show_hints): use proper state-to-string function for hint state (issue #5583) - - * rtp.c: use unsigned format for debug packet output (issue #5595) - - * asterisk.c (main): force a dnsmgr background refresh after all other modules are initialized (issue #5599) - * dnsmgr.c: add ability to start a background refresh on demand (issue #5599) - - * apps/app_dial.c (HANDLE_CAUSE): set CDR disposition to match cause code (issue #5602) - - * asterisk.c: support 'runuser' and 'rungroup' options in asterisk.conf (issue #5621) - - * res/Makefile, apps/Makefile, channels/Makefile, Makefile: support WITHOUT_ZAPTEL define to forcibly avoid building Zaptel support (issue #5634) - - * Makefile: various fixes (issue #5633) - - * apps/app_osplookup.c: upgrade to new arg/option API and implement priority jumping control - - * channels/chan_misdn.c: various fixes (issue #5639) - * channels/misdn/isdn_lib.c: various fixes (issue #5639) - - * apps/app_playback.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_privacy.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_sendtext.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_transfer.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_txtcidname.c: upgrade to new arg/option API and implement priority jumping control - - * Makefile: restore function of 'dont-optimize' - - * config.c (config_text_file_load): don't generate log message when stat() fails - - * many files: clean up application documentation to not refer to return values, since they cannot be used in the dialplan (work done by Neil Lewis) - -2005-11-06 Russell Bryant <russell@digium.com> - - * many files: alphabetize options in applicaiton descriptions - - * channels/chan_iax2.c: Use an enum to define iax peer/user flags as well as the pvt structure state. Use the ast_flags macros for checking or setting the state. - - * sounds.txt: Add missing words from the description of the vm-opts prompt - - * apps/app_externalivr.c: Add a space that fixes building on older versions of gcc - - * many files: Add doxygen updates to categorize modules into groups. Convert a lot of comments over to doxygen style. Add some text giving a basic overview of channels. - - * many files: Update applications to add an exit status variable, make priority jumping optional, and use new args parsing macros - - * pbx.c cdr.c res/res_features.c apps/app_dial.c include/asterisk/cdr.h: Convert some built-in applications to use new args parsing macros. Change ast_cdr_reset to take a pointer to an ast_flags structure instead of an integer for flags. - - * channels/chan_agent.c: Don't loop forever on an invalid options string - - * apps/app_disa.c apps/app_forkcdr.c: Fix to use correct arguments to ast_cdr_reset - -2005-11-05 Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't rebuild asterisk/build.h unless the asterisk binary is going to be relinked for some other reason (stops spurious recompile/link every time 'make' is issued); clean up variable substitutions to use consistent syntax - * asterisk.c: don't include asterisk/build.h (it's unnecessary) - * cli.c: don't include asterisk/build.h, use extern references to buildinfo.c - * buildinfo.c: new file to hold version info strings - -2005-11-04 Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_mixmonitor.c (mixmonitor_exec): correct app name in an error message - -2005-11-04 Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Create a function that stores a peer's status in a given buffer. Use this function in "iax2 show peers" and "iax2 show peer <peername>". Also, add the peer's status as an option to the IAXPEER dialplan function. - -2005-11-04 Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/compiler.h: don't try to use always_inline on old compilers - -2005-11-03 Russell Bryant <russell@digium.com> - - * res/res_agi.c: initialize buffer for result so that the contents are always valid in the response to GET FULL VARIABLE - -2005-11-03 Kevin P. Fleming <kpfleming@digium.com> - - * doc/README.variables: document DYNAMIC_FEATURES - - * res/res_features.c (ast_bridge_call): remove unused variables - - * apps/app_dial.c (dial_exec_full): simplify options and flag usage - - * include/asterisk/app.h: re-work application arg/option parsing APIs for consistent naming, add doxygen docs for option API - * many files: update to new APIs - -2005-11-02 Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c (dial_exec_full): convert to use API calls for argument/option parsing - - * include/asterisk/channel.h: add doxygen docs for silence generator APIs - - * channel.c (ast_channel_bridge): simplify native-bridge return logic, remove 'unsuccessful' message since it causes too many questions :-) - -2005-11-01 Kevin P. Fleming <kpfleming@digium.com> - - * stdtime/localtime.c: fix build failure on uClibc systems (issue #5558) - * devicestate.c: same - - * many files: make chan_misdn actually build (issue #5566) - - * many files: more Cygwin build system support (issue #4678) - - * apps/app_parkandannounce.c (parkandannounce_exec): supply parent channel to ast_request_and_dial so channel variables can be inherited (issue #5564) - * include/asterisk/channel.h: add parent_channel field - * channel.c (__ast_request_and_dial): use parent_channel field to inherit variables into new channel - - * apps/app_cut.c (cut_internal): use ast_app_separate_args() instead of open code (issue #5560) - - * apps/app_mixmonitor.c (launch_monitor_thread): ast_strlen_zero can handle NULL input (issue #5561) - (mixmonitor_exec): same - - * res/res_features.c (ast_feature_request_and_dial): ensure that channel variables are inherited from the channel placing the call (issue #5499) - - * utils.c (getloadavg): change to using _BSD_SOURCE as the indicator for whether this function is present or not (issue #5549) - - * include/asterisk/utils.h (ast_slinear_saturated_add): force to be inlined whenever possible - (ast_slinear_saturated_multiply): same - (ast_slinear_saturated_divide): same - (inaddrcmp): same - * include/asterisk/strings.h (ast_strlen_zero): force to be inlined whenever possible - * include/asterisk/compiler.h (force_inline): add macro to force inlining of functions - - * app.c (ast_play_and_record): use ast_silence_generator during recording if requested - * asterisk.c: add global option to enable silence-during-record (issue #5135) - * channel.c (silence_generator_alloc): new - (silence_generator_release): new - (silence_generator_generate): new - (ast_channel_start_silence_generator): new API call to start generating silence on a channel - (ast_channel_stop_silence_generator): parallel call to stop silence generation - * apps/app_record.c (record_exec): use ast_silence_generator during recording if requested - -2005-11-01 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0-beta2 released. - diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c index e4d826798..820f474bd 100644 --- a/channels/chan_iax2.c +++ b/channels/chan_iax2.c @@ -2620,7 +2620,7 @@ static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, static struct iax2_peer *realtime_peer(const char *peername, struct sockaddr_in *sin) { - struct ast_variable *var; + struct ast_variable *var = NULL; struct ast_variable *tmp; struct iax2_peer *peer=NULL; time_t regseconds, nowtime; @@ -2654,7 +2654,7 @@ static struct iax2_peer *realtime_peer(const char *peername, struct sockaddr_in * is because we only have the IP address and the host field might be * set as a name (and the reverse PTR might not match). */ - if (var) { + if (var && sin) { for (tmp = var; tmp; tmp = tmp->next) { if (!strcasecmp(tmp->name, "host")) { struct in_addr sin2 = { 0, }; |