diff options
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 22076 | ||||
| -rw-r--r-- | apps/app_userevent.c | 11 | ||||
| -rw-r--r-- | apps/app_voicemail.c | 246 | ||||
| -rw-r--r-- | channels/chan_local.c | 8 | ||||
| -rw-r--r-- | channels/chan_sip.c | 6 | ||||
| -rw-r--r-- | include/asterisk/say.h | 4 | ||||
| -rw-r--r-- | main/manager.c | 8 | ||||
| -rw-r--r-- | main/say.c | 109 | ||||
| -rw-r--r-- | res/res_features.c | 26 |
11 files changed, 248 insertions, 22248 deletions
diff --git a/.lastclean b/.lastclean deleted file mode 100644 index bb95160cb..000000000 --- a/.lastclean +++ /dev/null @@ -1 +0,0 @@ -33 diff --git a/.version b/.version deleted file mode 100644 index 580ef4b6c..000000000 --- a/.version +++ /dev/null @@ -1 +0,0 @@ -1.4.23 diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index c4f174bdc..000000000 --- a/ChangeLog +++ /dev/null @@ -1,22076 +0,0 @@ -2009-01-20 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.4.23 released. - -2009-01-16 00:19 +0000 [r168745] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump, - in this case) fails a consistency check because it can't find a - matching extension. The problem was a missing instruction to end - the range notation in the code where it converts the pattern into - a regex and uses the regex code to determine the match. I tested - using the AEL code the user supplied, and now, the consistency - check passes. (closes issue #14141) Reported by: dimas - -2009-01-15 18:43 +0000 [r168721] Olle Johansson <oej@edvina.net> - - * configs/extconfig.conf.sample: Meetme actually has realtime but - wasn't documented - -2009-01-15 18:22 +0000 [r168716] Terry Wilson <twilson@digium.com> - - * res/res_features.c: Convert call to park_call_full to - masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE - return value, we need to use masqueraded parking, otherwise we - will try to call ast_hangup() in __pbx_run() and in - do_parking_thread() and then promptly crash. (closes issue - #14215) Reported by: waverly360 Tested by: otherwiseguy (closes - issue #14228) Reported by: kobaz Tested by: otherwiseguy - -2009-01-15 01:20 +0000 [r168633] Tilghman Lesher <tlesher@digium.com> - - * /: Blocked revision 168632 from /branches/1.2: 1.2 regression on - security fix AST-2009-001 (Closes issue #14238) - -2009-01-15 00:11 +0000 [r168628] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix some crashes from bad datastore handling in - app_queue.c * The queue_transfer_fixup function was searching for - and removing the datastore from the incorrect channel, so this - was fixed. * Most datastore operations regarding the - queue_transfer datastore were being done without the channel - locked, so proper channel locking was added, too. (closes issue - #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by - putnopvut (license 60) Tested by: ZX81, festr - -2009-01-14 21:48 +0000 [r168622] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c: * Fixed create_process() allocation of - process ID values. The allocated process IDs could overflow their - respective NT and TE fields. Affects outgoing calls. - -2009-01-14 20:52 +0000 [r168614] Sean Bright <sean.bright@gmail.com> - - * contrib/scripts/autosupport: Update autosupport script to supply - info for both Zaptel and DAHDI in 1.4 and be sure to run - dahdi_test in 1.6.x and trunk instead of zttest. (closes issue - #14132) Reported by: dsedivec Patches: - asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638) - asterisk-trunk-autosupport.patch uploaded by dsedivec (license - 638) - -2009-01-14 19:34 +0000 [r168608] Steve Murphy <murf@digium.com> - - * apps/app_page.c: app_page was failing to compile in dev-mode on - my gcc-4.2.4 system. This change gets rid of the warning. - -2009-01-14 19:02 +0000 [r168603] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c: Don't read into a buffer without first checking if - a value is beyond the end. (closes issue #13600) Reported by: - atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 - (license 14) Tested by: atis - -2009-01-14 16:19 +0000 [r168598] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Fix a logic error I found while searching - through chan_agent.c I found that the allow_multiple_logins - function would never return 0 due to an incorrect comparison - being used when traversing the list of agents. While I was - modifying this function, I also did a little bit of coding - guidelines cleanup, too. - -2009-01-14 01:27 +0000 [r168593] Terry Wilson <twilson@digium.com> - - * apps/app_page.c: Don't overflow when paging more than 128 - extensions The number of available slots for calls in app_page - was hardcoded to 128. Proper bounds checking was not in place to - enforce this limit, so if more than 128 extensions were passed to - the Page() app, Asterisk would crash. This patch instead - dynamically allocates memory for the ast_dial structures and - removes the (non-functional) arbitrary limit. This issue would - have special importance to anyone who is dynamically creating the - argument passed to the Page application and allowing more than - 128 extensions to be added by an outside user via some external - interface. The patch posted by a_villacis was slightly modified - for some coding guidelines and other cleanups. Thanks, - a_villacis! (closes issue #14217) Reported by: a_villacis - Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch - uploaded by a (license 660) Tested by: otherwiseguy - -2009-01-13 19:13 +0000 [r168561] Russell Bryant <russell@digium.com> - - * main/indications.c, main/channel.c, apps/app_read.c, - channels/chan_misdn.c, funcs/func_channel.c, - include/asterisk/indications.h, apps/app_disa.c, main/app.c, - res/snmp/agent.c, include/asterisk/channel.h, - res/res_indications.c: Revert unnecessary indications API change - from rev 122314 - -2009-01-13 18:34 +0000 [r168551] Terry Wilson <twilson@digium.com> - - * channels/chan_sip.c: Don't pass a value with a side effect to a - macro (closes issue #14176) Reported by: paraeco Patches: - chan_sip.c.diff uploaded by paraeco (license 658) - -2009-01-13 17:48 +0000 [r168546] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_logic.c: If either conditional is NULL, don't try - copying it. (closes issue #14226) Reported by: caspy Patches: - 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) - -2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler <jpeeler@digium.com> - - * res/res_agi.c: (closes issue #13881) Reported by: hoowa Update - the app CDR field for AGI commands that are not executing an - application via "exec". - - * channels/chan_agent.c: (closes issue #12269) Reported by: IgorG - Tested by: denisgalvao This gits rid of the notion of an - owning_app allowing the request and hangup to be initiated by - different threads. Originating from an active agent channel - requires this. The implementation primarily changes __login_exec - to wait on a condition variable rather than a lock. Review: - http://reviewboard.digium.com/r/35/ - -2009-01-12 14:58 +0000 [r168482] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: I am reverting the fix made in revision - 168128 (and its upward merges) after being contacted by Olle - Johansson and being shown how this fix is incorrect. Thanks to - Olle for clearing this up for me. - -2009-01-12 14:57 +0000 [r168480] Russell Bryant <russell@digium.com> - - * configs/indications.conf.sample: s/ringdance/ringcadence/ for - Bulgaria - -2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming <kpfleming@digium.com> - - * README: small commit to test new server - - * README: small commit to test new server - - * sounds/Makefile: update to use new sound file packages that - include license files - -2009-01-09 22:14 +0000 [r168198] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Make this compile for mvanbaak - -2009-01-09 21:28 +0000 [r168191] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 * - Miscellaneous doxygen comments added. - -2009-01-09 20:08 +0000 [r168128] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Add check_via calls to more request handlers - INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not - checking the topmost Via to determine where to send the response. - Adding check_via calls to those request handlers solves this. - (closes issue #13071) Reported by: baron Patches: check_via.patch - uploaded by baron (license 531) Tested by: baron - -2009-01-08 22:08 +0000 [r167840] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Don't truncate database results at 255 chars. - (closes issue #14069) Reported by: evandro Patches: - 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) - -2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: remove an unnecessary argument to - queue_request() - - * channels/chan_sip.c: When a SIP request or response arrives for a - dialog with an associated Asterisk channel, and the lock on that - channel cannot be obtained because it is held by another thread, - instead of dropping the request/response, queue it for later - processing when the channel lock becomes available. - http://reviewboard.digium.com/r/117/ - -2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant <russell@digium.com> - - * main/file.c: Fix the last couple of places where free() was - improperly used directly. - - * main/file.c: Don't fclose() the file early, the filestream - destructor will handle it. - - * main/file.c: Only try to close the file if one was actually - opened - - * main/file.c: Don't use free() directly. This caused a crash since - ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, - #asterisk-dev - - * main/indications.c: Treat an empty string the same way as a NULL - country argument. In passing, simplify the handling of returning - a default tone zone. - -2009-01-06 21:35 +0000 [r167299] Mark Michelson <mmichelson@digium.com> - - * main/db.c: Use the correct variable when creating the format - string (closes issue #14177) Reported by: nic_bellamy Patches: - asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic - (license 299) - -2009-01-06 20:48 +0000 [r167260] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 - Jan 2009) | 2 lines Security fix AST-2009-001. ........ - -2009-01-05 16:51 +0000 [r167179] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: A couple of changes to T.38 SDP attribute - handling There are some boolean attributes for T.38 such as - T38FaxFillBitRemoval, T38FaxTranscodingMMR, and - T38FaxTranscodingJBIG. By simply being present, we should treat - these as a "true" value. The current code, however, was requiring - a 1 or 0 as the value of the attribute in order to parse it. This - is due to the fact that there are some T.38 endpoints and - gateways that also transmit this information incorrectly. This - patch follows the "be liberal in what you accept and strict in - what you send" philosophy by accepting both the correctly- and - incorrectly-formatted attributes, but only sending information as - it is supposed to be sent. It was also discovered that a - particular type of T.38 gateway sends some non-standard T.38 SDP - attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, - it used T38MaxDatagram and T38FaxMaxRate respectively. We now - will properly accept these attributes as well. Note that there - are a lot of patches cited in the below commit message template. - This is because the person who submitted these patches is an - awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes - issue #13976) Reported by: linulin Patches: - chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) - chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) - chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) - chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov - (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded - by arcivanov (license 648) - chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov - (license 648) Tested by: arcivanov - -2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_alsa.c: Repeat attempts to write when we receive - -EAGAIN from the driver, as detailed in the ALSA sample code (see - http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) - Reported by: Jerry Geis (via the -users list) Fixed by: me - (license 14) - - * channels/chan_local.c: Also inherit the musiconhold class. - (Closes #14153) Reported by: Jerry Geis, via the users list. - Patch by: me (license 14) - -2008-12-28 15:13 +0000 [r166772] Russell Bryant <russell@digium.com> - - * channels/misdn_config.c: Use strncat() instead of an sprintf() in - which source and target buffers overlap - http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html - -2008-12-23 15:35 +0000 [r166592] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, channels/chan_iax2.c: Compile, even if both - DAHDI and Zaptel are not installed. (Closes issue #14120) - -2008-12-23 15:16 +0000 [r166568] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: Fix a crash resulting from a datastore with - inheritance but no duplicate callback The fix for this is to - simply set the newly created datastore's data pointer to NULL if - it is inherited but has no duplicate callback. (closes issue - #14113) Reported by: francesco_r Patches: 14113.patch uploaded by - putnopvut (license 60) Tested by: francesco_r - -2008-12-23 04:05 +0000 [r166509] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c: Use the integer form of condition for integer - comparisons. (closes issue #14127) Reported by: andrew - -2008-12-22 20:56 +0000 [r166380] Mark Michelson <mmichelson@digium.com> - - * channels/chan_dahdi.c: Fix a deadlock relating to channel locks - and autoservice It has been discovered that if a channel is - locked prior to a call to ast_autoservice_stop, then it is likely - that a deadlock will occur. The reason is that the call to - ast_autoservice_stop has a check built into it to be sure that - the thread running autoservice is not currently trying to - manipulate the channel we are about to pull out of autoservice. - The autoservice thread, however, cannot advance beyond where it - currently is, though, because it is trying to acquire the lock of - the channel for which autoservice is attempting to be stopped. - The gist of all this is that a channel MUST NOT be locked when - attempting to stop autoservice on the channel. In this particular - case, the channel was locked by a call to ast_read. A call to - ast_exists_extension led to autoservice being started and stopped - due to the existence of dialplan switches. It may be that there - are future commits which handle the same symptoms but in a - different location, but based on my looks through the code, it is - very rare to see a construct such as this one. (closes issue - #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded - by putnopvut (license 60) Tested by: rtrauntvein Review: - http://reviewboard.digium.com/r/107/ - -2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant <russell@digium.com> - - * main/utils.c: Fix up timeout handling in ast_carefulwrite(). - - * include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref - count handling of MoH classes using astobj2 to resolve crashes. - (closes issue #13566) Reported by: igorcarneiro Tested by: - russell Review: http://reviewboard.digium.com/r/106/ - -2008-12-19 23:34 +0000 [r166157] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, funcs/func_audiohookinherit.c (added), - channels/chan_sip.c, include/asterisk/audiohook.h, - main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for - Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches: - 13538.patch uploaded by putnopvut (license 60) Tested by: - putnopvut - -2008-12-19 22:30 +0000 [r166093] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h, - apps/app_queue.c: This merges the masqpark branch into 1.4 These - changes eliminate the need for (and use of) the KEEPALIVE return - code in res_features.c; There are other places that use this - result code for similar purposes at a higher level, these appear - to be left alone in 1.4, but attacked in trunk. The reason these - changes are being made in 1.4, is that parking ends a channel's - life, in some situations, and the code in the bridge (and some - other places), was not checking the result code properly, and - dereferencing the channel pointer, which could lead to memory - corruption and crashes. Calling the masq_park function eliminates - this danger in higher levels. A series of previous commits have - replaced some parking calls with masq_park, but this patch puts - them ALL to rest, (except one, purposely left alone because a - masquerade is done anyway), and gets rid of the code that tests - the KEEPALIVE result, and the NOHANGUP_PEER result codes. While - bug 13820 inspired this work, this patch does not solve all the - problems mentioned there. I have tested this patch (again) to - make sure I have not introduced regressions. Crashes that - occurred when a parked party hung up while the parking party was - listening to the numbers of the parking stall being assigned, is - eliminated. These are the cases where parking code may be - activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to - parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip - xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via - manager. The interesting testing cases for parking are: I. A - calls B, A parks B a. B hangs up while A is getting the numbers - announced. b. B hangs up after A gets the announcement, but - before the parking time expires c. B waits, time expires, A is - redialed, A answers, B and A are connected, after which, B hangs - up. d. C picks up B while still in parking lot. II. A calls B, B - parks A a. A hangs up while B is getting the numbers announced. - b. A hangs up after B gets the announcement, but before the - parking time expires c. A waits, time expires, B is redialed, B - answers, A and B are connected, after which, A hangs up. d. C - picks up A while still in parking lot. Testing this throroughly - involves acting all the permutations of I and II, in situations - 1,2,3, and 4. Since I added a few more changes (ALL references to - KEEPALIVE in the bridge code eliimated (I missed one earlier), I - retested most of the above cases, and no crashes. H-extension - weirdness. Current h-extension execution is not completely - correct for several of the cases. For the case where A calls B, - and A parks B, the 'h' exten is run on A's channel as soon as the - park is accomplished. This is expected behavior. But when A calls - B, and B parks A, this will be current behavior: After B parks A, - B is hung up by the system, and the 'h' (hangup) exten gets run, - but the channel mentioned will be a derivative of A's... Thus, if - A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on - channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info - will be those relating to Channel A. And, in the case where A is - reconnected to B after the park time expires, when both parties - hang up after the joyful reunion, no h-exten will be run at all. - In the case where C picks up A from the parking lot, when either - A or C hang up, the h-exten will be run for the C channel. CDR's - are a separate issue, and not addressed here. As to WHY this - strange behavior occurs, the answer lies in the procedure - followed to accomplish handing over the channel to the parking - manager thread. This procedure is called masquerading. In the - process, a duplicate copy of the channel is created, and most of - the active data is given to the new copy. The original channel - gets its name changed to XXX<ZOMBIE> and keeps the PBX - information for the sake of the original thread (preserving its - role as a call originator, if it had this role to begin with), - while the new channel is without this info and becomes a call - target (a "peer"). In this case, the parking lot manager thread - is handed the new (masqueraded) channel. It will not run an - h-exten on the channel if it hangs up while in the parking lot. - The h exten will be run on the original channel instead, in the - original thread, after the bridge completes. See bug 13820 for - our intentions as to how to clean up the h exten behavior. - Review: http://reviewboard.digium.com/r/29/ - -2008-12-19 19:48 +0000 [r165991] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c, - apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c, - apps/app_dahdiscan.c, codecs/codec_dahdi.c, - res/res_musiconhold.c, channels/chan_iax2.c: (closes issue - #13480) Reported by: tzafrir Replace a bunch of if defined checks - for Zaptel/DAHDI through several new defines in dahdi_compat.h. - This removes a lot of code duplication. Example from bug: #ifdef - HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd = - open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd = - open(DAHDI_FILE_PSEUDO, O_RDRW); - -2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c: Ensure that the chanspy datastore is fully - initialized. This patch resolved some random crash issues - observed by a user on a BSD system (closes issue #14111) Reported - by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) - - * main/utils.c: Make ast_carefulwrite() be more careful. This patch - handles some additional cases that could result in partial writes - to the file description. This was done to address complaints - about partial writes on AMI. (issue #13546) (more changes needed - to address potential problems in 1.6) Reported by: srt Tested by: - russell Review: http://reviewboard.digium.com/r/99/ - -2008-12-18 21:14 +0000 [r165767] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Add mutexes around accesses to the IMAP - library interface. This prevents certain crashes, especially when - shared mailboxes are used. (closes issue #13653) Reported by: - howardwilkinson Patches: - asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by - howardwilkinson (license 590) Tested by: jpeeler - -2008-12-18 18:52 +0000 [r165661] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Set the process group ID on the MOH - process so that all children will get killed (closes issue - #14099) Reported by: caspy Patches: - res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license - 645) - -2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Only care about a compatible codec for early bridging - if we are actually bridging to another channel. If we are not we - actually want to bring the audio back to us. (closes issue - #13545) Reported by: davidw - - * apps/app_followme.c: Do not crash if we are not passed in a - followme id. (closes issue #14106) Reported by: ys Patches: - app_followme.c.2.diff uploaded by ys (license 281) - -2008-12-17 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.23-rc3 released. - -2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher@digium.com> - - * apps/app_macro.c: Reverse the fix from issue #6176 and add proper - handling for that issue. (Closes issue #13962, closes issue - #13363) Fixed by myself (license 14) - -2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson@digium.com> - - * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c, - apps/app_queue.c: Fix some memory leaks found while looking at - how realtime configs are handled. Also cleaned up some coding - guidelines violations in app_realtime.c, mostly related to - spacing - - * channels/chan_sip.c: After looking through SIP registration code - most of the day, this is one of the few things I could find that - was just plain wrong. Even though it probably isn't possible for - it to happen, it seems weird to have code that checks if a - pointer is NULL and then immediately dereferences that pointer if - it was NULL. - -2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell@digium.com> - - * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously - report that a thread is exiting while holding a lock. If the last - lock attempt was a trylock, and it failed, it will still be in - the list of locks so that it can be reported. (closes issue - #13219) Reported by: pj - - * apps/app_macro.c: Do not dereference the channel if - AST_PBX_KEEPALIVE has been returned. This is a bug I noticed - while looking at the code for app_macro. This return code means - that another thread has assumed ownership of the channel and it - can no longer be touched. (I hate this return code with a - passion, by the way.) - - * main/manager.c: Add "restart gracefully" to the AMI blacklist of - CLI commands. "module unload" was already identified as a command - that can not be used from the AMI. "restart gracefully" - effectively unloads all modules, and will run in to the same - problems. (closes issue #13894) Reported by: kernelsensei - - * include/asterisk/threadstorage.h, main/threadstorage.c: Fix - memory leak and invalid reporting issues with DEBUG_THREADLOCALS. - One issue was that the ast_mutex_* API was being used within the - context of the thread local data destructors. We would go off and - allocate more thread local data while the pthread lib was in the - middle of destroying it all. This led to a memory leak. Another - issue was an invalid argument being provided to the the - object_add API call. (closes issue #13678) Reported by: ys Tested - by: Russell - - * channels/chan_sip.c: Fix a memory leak related to the use of the - "setvar" configuration option. The problem was that these - variables were being appended to the list of vars on the sip_pvt - every time a re-registration or re-subscription came in. Since - it's just a waste of memory to put them there unless the request - was an INVITE, then the fix is to check the request type before - copying the vars. (closes issue #14037) Reported by: marvinek - Tested by: russell - -2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf@digium.com> - - * main/pbx.c: I added a sentence to clarify why - and ' ' are - ignored in patterns as per bug 14076. Leif says he'll put some - stuff about it in the extensions.conf sample, etc. - -2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Don't try to change working directory if a - directory was not configured. (closes issue #14089) Reported by: - caspy - -2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/pbx.h: Add the deadlock note to - ast_spawn_extension as well - - * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to - autoservice and pbx doxygen regarding a potential deadlock - scenario so that it is avoided in the future - -2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Do not try to unlock a non-existant channel - if the transfer fails. (closes issue #13800) Reported by: dwagner - Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety - (license 608) - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/channel.h: Use autoconf logic to determine - whether the system has timersub or not. Do not blindly assume - Solaris does not. (closes issue #13838) Reported by: ano - - * apps/app_dial.c: Can we try not to assign an unsigned int to -1? - (closes issue #14074) Reported by: wetwired - -2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell@digium.com> - - * main/channel.c, res/res_features.c: Handle a case where a call - can be bridged to a channel that is still ringing. The issue that - was reported was about a case where a RINGING channel got - redirected to an extension to pick up a call from parking. Once - the parked call got taken out of parking, it heard silence until - the other side answered. Ideally, the caller that was parked - would get a ringing indication. This patch fixes this case so - that the caller receives ringback once it comes out of parking - until the other side answers. The fixes are: - Make sure we - remember that a channel was an outgoing channel when doing a - masquerade. This prevents an erroneous ast_answer() call on the - channel, which causes a bogus 200 OK to be sent in the case of - SIP. - Add some additional comments to explain related parts of - code. - Update the handling of the ast_channel visible_indication - field. Storing values that are not stateful is pointless. Control - frames that are events or commands should be ignored. - When a - bridge first starts, check to see if the peer channel needs to be - given ringing indication because the calling side is still - ringing. - Rework ast_indicate_data() a bit for the sake of - readability. (closes issue #13747) Reported by: davidw Tested by: - russell Review: http://reviewboard.digium.com/r/90/ - -2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c: Change the default calldurationlimit from the - special value 0 to -1, so we can better detect an exceptional - case. This follows on to the changes made in revision 156386. - Related to issue #13851. (closes issue #13974) Reported by: - paradise Patches: 20081208__bug13974.diff.txt uploaded by - Corydon76 (license 14) Tested by: file, blitzrage, ZX81 - -2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell@digium.com> - - * /: Set the reviewboard:url property on 1.4, as well - -2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not - immediately exiting Asterisk, but also add a pointer inside - editline to look back to asterisk.c, so others don't spend as - much time as I did looking (in the wrong place) for the - appropriate function. Reported by: ZX81, via the #asterisk-users - channel Fixed by: me (license 14) - -2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32 - so that it is actually 32 bits on 64-bit machines, as well. - (inspired by issue #13879) - - * main/channel.c, main/autoservice.c, include/asterisk/channel.h: - Resolve issues that could cause DTMF to be processed out of - order. These changes come from team/russell/issue_12658 1) Change - autoservice to put digits on the head of the channel's frame - readq instead of the tail. If there were frames on the readq that - autoservice had not yet read, the previous code would have - resulted in out of order processing. This required a new API call - to queue a frame to the head of the queue instead of the tail. 2) - Change up the processing of DTMF in ast_read(). Some of the - problems were the result of having two sources of pending DTMF - frames. There was the dtmfq and the more generic readq. Both were - used for pending DTMF in various scenarios. Simplifying things to - only use the frame readq avoids some of the problems. 3) Fix a - bug where a DTMF END frame could get passed through when it - shouldn't have. If code set END_DTMF_ONLY in the middle of digit - emulation, and a digit arrived before emulation was complete, - digits would get processed out of order. (closes issue #12658) - Reported by: dimas Tested by: russell, file Review: - http://reviewboard.digium.com/r/85/ - -2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console, - on certain shells, the terminal is messed up. By intercepting - those events with a signal handler in the remote console, we can - avoid those issues. (closes issue #13464) Reported by: tzafrir - Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 - (license 14) Tested by: blitzrage - -2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson@digium.com> - - * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes. - (closes issue #13819) Reported by: adomjan Patches: - pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) - dundi_clearecache3.diff uploaded by mnicholson (license 96) - Tested by: adomjan - -2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell@digium.com> - - * funcs/func_strings.c, funcs/func_cut.c: Fix some observed - slowdowns in dialplan processing. The change is to remove - autoservice usage from dialplan functions that do not need it - because they do not perform operations that potentially block. - (closes issue #13940) Reported by: tbelder - - * res/res_features.c: Fix an issue that made it so you could only - have a single caller executing a custom feature at a time. This - was especially problematic when custom features ran for any - appreciable amount of time. The fix turned out to be quite - simple. The dynamic features are now stored in a read/write list - instead of a list using a mutex. (closes issue #13478) Reported - by: neutrino88 Fix suggested by file - -2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Don't wait forever, if there's a specified - recording timeout. (closes issue #13885) Reported by: bamby - Patches: res_agi.c.patch uploaded by bamby (license 430) - -2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Revert this cast to long. Using time_t here - causes build failures on a FreeBSD 32-bit build. - - * apps/app_queue.c: Fix a potential crash due to unsafe datastore - handling. This patch also contains a conversion from using long - to time_t for representing times for a queue, as well as some - whitespace fixes. (closes issue #14060) Reported by: nivek - Patches: datastore_fixup.patch.corrected uploaded by nivek - (license 636) with slight modification from me Tested by: nivek - -2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler@digium.com> - - * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp - check. Pointed out by mmichelson, thanks! - - * res/res_musiconhold.c: (closes issue #13229) Reported by: - clegall_proformatique Ensure that moh_generate does not return - prematurely before local_ast_moh_stop is called. Also, the sleep - in mp3_spawn now only occurs for http locations since it seems to - have been added originally only for failing media streams. - -2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix subscription based MWI up a bit. We only - want to put sip: at the beginning of the URI if it is not already - there and revert code to ignore destination check if subscribing - for MWI. (closes issue #12560) Reported by: vsauer Patches: - patch001.diff uploaded by ramonpeek (license 266) - - * channels/chan_sip.c: When a SIP peer unregisters set the expiry - time back to 0 so that the 200 OK contains an expires of 0. - (closes issue #13599) Reported by: hjourdain Patches: - chan_sip.c.diff uploaded by hjourdain (license 583) - -2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf@digium.com> - - * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022) - Reported by: wetwired Tested by: murf I checked, and I added a - mod to the trunk version of Asterisk that would make it 8-bit - transparent on 27 Nov 2007, but I made no such updates to 1.4. My - best guess is that 1.4 was released, and it was not appropriate - to commit an enhancement. But I'm going to add the same fixes to - 1.4 now, for the following reasons: 1. wetwired is correct; 1.4 - is **mostly** 8-bit transparent now. This is because the lexical - token forming rules use . in most 'word' state continuances. It's - just the beginning of a 'word' that is picky. 2. Accepting 8-bit - chars in some places and not others leads to bug reports like - this. - -2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/stringfields.h: Update to stringfield handling - so that side-effects on parameters are not evaluated multiple - times. An example where this caused a problem was in chan_sip.c, - with the line ast_string_field_set(p, fromdomain, ++fromdomain); - This patch was originally uploaded to issue #13783 by jamessan. - While the issue was closed for other reasons, this patch is valid - and fixes a separate problem, and is thus being committed. - - * channels/chan_sip.c: Revert fix for issue 13570. It has caused - more problems than it helped to fix. (closes issue #13783) - Reported by: navkumar (closes issue #14025) Reported by: ffs - - * doc/misdn.txt: Add missing documentation to misdn.txt (closes - issue #14052) Reported by: festr Patches: misdn.txt.patch - uploaded by festr (license 443) - -2008-12-10 16:05 +0000 [r162653] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Increment the sequence number on the end packets for - RFC2833. After reading the RFC some more and doing some testing I - agree with this change. (closes issue #12983) Reported by: vt - Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license - 520) - -2008-12-09 23:08 +0000 [r162463] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Oops, should be "tz", not "zonetag". - -2008-12-09 22:17 +0000 [r162413] Russell Bryant <russell@digium.com> - - * main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove - the test_for_thread_safety() function completely. The test is not - valid. Besides, if we actually suspected that recursive mutexes - were not working, we would get a ton of LOG_ERROR messages when - DEBUG_THREADS is turned on. (inspired by a discussion on the - asterisk-dev list) - -2008-12-09 21:53 +0000 [r162348] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: We appear to have documented tz= in the - [general] section of voicemail.conf, without actually having - implemented it. Oops. (Reported by Olivier on the -users list) - -2008-12-09 21:14 +0000 [r162341] Joshua Colp <jcolp@digium.com> - - * apps/app_directed_pickup.c: Add 'down' as a valid state for - directed call pickup. This creeps up when we receive session - progress when dialing a device and not ringing. (closes issue - #14005) Reported by: ddl - -2008-12-09 20:57 +0000 [r162286] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Fix an issue where callers on an incoming call - on an SLA trunk would not hear ringback. We need to make sure - that we don't start writing audio to the trunk channel until - we're actually ready to answer it. Otherwise, the channel driver - will treat it as inband progress, even though all they are - getting is silence. (closes issue #12471) Reported by: mthomasslo - -2008-12-09 20:44 +0000 [r162273] Joshua Colp <jcolp@digium.com> - - * apps/app_festival.c: Fix double declaration of 'x' on the PPC - platform. (closes issue #14038) Reported by: ffloimair - -2008-12-09 20:28 +0000 [r162265] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: If we fail to start a thread for the pbx to run in, - we need to be sure to decrease the number of active calls on the - system. This fix may relate to ABE-1713, but it is not certain - yet. - -2008-12-09 20:20 +0000 [r162264] Steve Murphy <murf@digium.com> - - * pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with - seanbright on #asterisk-dev, I have added a default rule, and an - option to suppress the default rule from being generated in the - flex output, for the sake of those OS's where they didn't tweak - flex's ECHO macro, and the compiler doesn't like it. The - regressions are OK with this. - -2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Make sure that the timestamp for DTMF is not the same - as the previous voice frame and do not send audio when - transmitting DTMF as this confuses some equipment. (closes issue - #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file - (license 11) Tested by: ip-rob, bujones - - * main/rtp.c: Take video into account when early bridging RTP. - (closes issue #13535) Reported by: davidw - -2008-12-09 18:13 +0000 [r162136] Steve Murphy <murf@digium.com> - - * pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc - and ast_copy_string and messed up the standalone stuff. Fixed. - -2008-12-09 17:07 +0000 [r162071] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_phone.c: For some reason, after a distclean, gcc - started returning 'value computed is not used'. Fixing (for - --enable-dev-mode). - -2008-12-09 16:46 +0000 [r162014] Russell Bryant <russell@digium.com> - - * apps/app_disa.c: Allow DISA to handle extensions that start with - #. (closes issue #13330) Reported by: jcovert - -2008-12-09 16:31 +0000 [r162013] Steve Murphy <murf@digium.com> - - * pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h, - pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme - Patches: 14019.diff uploaded by murf (license 17) Tested by: - ckjohnsonme, murf This crash was the result of a few small errors - that would combine in 64-bit land to result in a crash. 32-bit - land might have seen these combine to mysteriously drop the args - to an application call, in certain circumstances. Also, in trying - to find this bug, I spotted a situation in the flex input, where, - in passing back a 'word' to the parser, it would allocate a - buffer larger than necessary. I changed the usage in such - situations, so that strdup was not used, but rather, an - ast_malloc, followed by ast_copy_string. I removed a field from - the pval struct, in u2, that was never getting used, and set in - one spot in the code. I believe it was an artifact of a previous - fix to make switch cases work invisibly with extens. And, for - goto's I removed a '!' from before a strcmp, that has been there - since the initial merging of AEL2, that might prevent the proper - target of a goto from being found. This was pretty harmless on - its own, as it would just louse up a consistency check for users. - Many thanks to ckjohnsonme for providing a simplified and - complete set of information about the bug, that helped - considerably in finding and fixing the problem. Now, to get - aelparse up and running again in trunk, and out of its "horribly - broken" state, so I can run the regression suite! - -2008-12-09 14:52 +0000 [r161948] Russell Bryant <russell@digium.com> - - * main/app.c: Fix a problem with GROUP() settings on a masquerade. - The previous code carried over group settings from the old - channel to the new one. However, it did nothing with the group - settings that were already on the new channel. This patch removes - all group settings that already existed on the new channel. I - have a more complicated version of this patch which addresses - only the most blatant problem with this, which is that a channel - can end up with multiple group settings in the same category. - However, I could not think of a use case for keeping any of the - group settings from the old channel, so I went this route for - now. (closes AST-152) - -2008-12-08 17:52 +0000 [r161725] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make the usereqphone option work again. - (closes issue #13474) Reported by: mmaguire Patches: - 20080912_bug13474.diff uploaded by mmaguire (license 571) - -2008-12-05 21:02 +0000 [r161426] Sean Bright <sean.bright@gmail.com> - - * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions - 161421 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec - 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned - int). (closes issue #14006) Reported by: alphaque Patches: - astobj2.h-patch uploaded by alphaque (license 259) (Slightly - modified by seanbright) ........ - -2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard <dhubbard@digium.com> - - * utils/smsq.c: kill a warning - -2008-12-05 14:12 +0000 [r161287] Russell Bryant <russell@digium.com> - - * main/pbx.c: Fix a NULL format string warning found by buildbot. - -2008-12-04 18:30 +0000 [r161013] Jeff Peeler <jpeeler@digium.com> - - * main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by: - jpeeler This mirrors a check that was present in ast_rtp_read to - also be in ast_rtp_raw_write to not schedule sending the receiver - report if the remote RTCP endpoint address isn't present in the - RTCP structure. Closes AST-142. - -2008-12-04 16:44 +0000 [r160943] Mark Michelson <mmichelson@digium.com> - - * main/callerid.c: Fix a callerid parsing issue. If someone - formatted callerid like the following: "name <number>" (including - the quotation marks), then the parts would be parsed as name: - "name number: number This is because the closing quotation mark - was not discovered since the number and everything after was - parsed out of the string earlier. Now, there is a check to see if - the closing quote occurs after the number, so that we can know if - we should strip off the opening quote on the name. Closes AST-158 - -2008-12-03 21:54 +0000 [r160770] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Some compilers warn on null format strings; - some don't (caught by buildbot) - -2008-12-03 21:38 +0000 [r160764] Jason Parker <jparker@digium.com> - - * channels/chan_agent.c: Only show this warning when we want to - show it. (closes issue #13982) Reported by: coolmig Patches: - chan_agent.c.patch uploaded by coolmig (license 621) - -2008-12-03 20:41 +0000 [r160703] Steve Murphy <murf@digium.com> - - * funcs/func_callerid.c: (closes issue #13597) Reported by: - john8675309 Patches: patch.13597 uploaded by murf (license 17) - Tested by: murf, john8675309 This patch causes the setcid func to - update the CDR clid after setting the channel field. - -2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: During bridge code, the channel bridge may - return a retry code, if a transfer was initiated but not yet - completed. If the bridge is immediately retried, then we may send - a storm of TXREQ packets, even though the first set is sent - reliably (retransmitted). Fixes AST-137. - - * pbx/pbx_spool.c: If an entry is added to the directory during a - scan when another entry expires, then that new entry will not be - processed promptly, but must wait for either a future entry to - start or a current entry's retry to occur. If no other entries - exist in the directory (other than the new entries) when a bunch - expire, then the new entries must wait until another new entry is - added to be processed. This was a rather weird race condition, - really. Fixes AST-147. - - * pbx/pbx_spool.c: Don't start scanning the directory until all - modules are loaded, because some required modules (channels, - apps, functions) may not yet be in memory yet. Fixes AST-149. - - * channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev - list: "I guess that having only ip-phones in mind is not a good - approach. Since it is possible to have a sip proxy connected to - asterisk we could receive a 407 (unauthorized) or 483 (too many - hops) as response and dialog ending would not be a good - behavior." So modified. - -2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson <twilson@digium.com> - - * res/res_features.c: Channel is masqueraded, don't keep alive - - * res/res_features.c: A situation like A calls B, A builtin_atxfers - B to C, C parks B would lead to a crash. Thanks to file for - telling me how to fix it! (closes issue #13854) Reported by: Adam - Lee Tested by: otherwiseguy - -2008-12-02 17:42 +0000 [r160297] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: When the text does not match exactly (e.g. - RTP/SAVP), then the %n conversion fails, and the resulting - integer is garbage. Thus, we must initialize the integer and - check it afterwards for success. (closes issue #14000) Reported - by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by - folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded - by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff - uploaded by folke (license 626) - -2008-12-02 01:16 +0000 [r160266] Terry Wilson <twilson@digium.com> - - * include/asterisk/astmm.h: make compile with dev mode and malloc - debug - -2008-12-02 00:25 +0000 [r160207] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/stringfields.h, apps/app_voicemail.c, - main/pbx.c, main/frame.c: Ensure that Asterisk builds with - --enable-dev-mode, even on the latest gcc and glibc. - -2008-12-01 Tilghman Lesher <tilghman@digium.com> - - * Released 1.4.23-rc2 - -2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to - iax2_setoption(), as well, since they both have the potential to - send control frames in the middle of call setup. We have to wait - until we have received a message back from the remote end before - we try to send any more frames. Otherwise, the remote end will - consider it invalid, and we'll get stuck in an INVAL/VNAK storm. - -2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel@vanbaak.info> - - * main/manager.c: Get rid of the useless format string and argument - in the Bogus/ manager channelname. Noted by kpfleming and name - Bogus/manager suggested by eliel - -2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell@digium.com> - - * .cleancount: Force a "make clean" to avoid a bizarre build issue - ... - -2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel@vanbaak.info> - - * main/manager.c: make manager compile on OpenBSD. The last (10th) - argument to ast_channel_alloc here should be a pointer and NULL - is not really a pointer. - -2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming@digium.com> - - * main/enum.c, utils/frame.c, configure, res/res_agi.c, - include/asterisk/module.h, main/logger.c, main/dns.c, - include/asterisk/threadstorage.h, include/asterisk/utils.h, - include/asterisk/devicestate.h, channels/chan_sip.c, - include/asterisk/dundi.h, main/jitterbuf.c, - channels/chan_agent.c, configure.ac, utils/astman.c, - include/asterisk/cli.h, include/asterisk/channel.h, - include/jitterbuf.h, include/asterisk/manager.h, - main/ast_expr2.c, Makefile, include/asterisk/logger.h, - include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c, - include/asterisk/linkedlists.h, include/asterisk/lock.h, - include/asterisk/strings.h, makeopts.in, - include/asterisk/stringfields.h, utils/check_expr.c, - channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c: - update dev-mode compiler flags to match the ones used by default - on Ubuntu Intrepid, so all developers will see the same warnings - and errors since this branch already had some printf format - attributes, enable checking for them and tag functions that - didn't have them format attributes in a consistent way - -2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_oss.c, channels/busy.h (removed), - channels/ring_tone.h (added), channels/chan_alsa.c, - channels/ringtone.h (removed), channels/busy_tone.h (added), - channels/Makefile: rename these files so as to avoid conflicts - when users update their working copies and have unversioned files - already in place - - * channels, agi/Makefile, utils/Makefile, channels/busy.h (added), - Makefile.moddir_rules, Makefile.rules, channels/ringtone.h - (added), channels/Makefile: simplify (and slightly bug-fix) the - recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h - and channels/ringtone.h to the repository instead of generating - them repeatedtly; most users do not change the settings to build - them, but the Makefile rules are still there if they wish to do - so ensure that 'make clean' removes dependency files for .i files - that are created in COMPILE_DOUBLE mode - -2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf@digium.com> - - * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported - by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) - Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX - fix. the change to cdr.c allows no-answer to percolate up into - CDR's, and feels like the right place to locate this fix; if BUSY - is done here, no-answer should be, too. - -2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Don't try to send a response on a NULL pvt. - (closes issue #13919) Reported by: barthpbx Patches: - chan_iax2.c.patch uploaded by eliel (license 64) Tested by: - barthpbx - - * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 - Nov 2008) | 7 lines Regression fix for last security fix. Set the - iseqno correctly. (closes issue #13918) Reported by: ffloimair - Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 - (license 14) Tested by: ffloimair ........ - -2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell@digium.com> - - * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to - go along with ao2_lock() and ao2_unlock() - -2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson@digium.com> - - * apps/app_festival.c: Add missing variable declaration in the PPC - code - -2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher@digium.com> - - * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in, - configure.ac: System call ioperm is non-portable, so check for - its existence in autoconf. (Closes issue #13863) - -2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes - issue #13786) Reported by: tzafrir When compiling against Zaptel - dahdi_compat will now only define all the DAHDI defines if the - Zaptel define is present. Also, there is no such thing as - DAHDI_PRI. - -2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf@digium.com> - - * res/res_features.c: In reference to the fix made for 13871, I was - merging the fix into 1.6.0 and realized I missed the code in the - h-exten block, and didn't catch it because my test case had the - h-exten commented out. So, this corrects the code I missed, as a - preventative against another crash report. Tested with the - h-exten defined, all is well. - -2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: The passed extension may not be the same in the list - as the current entry, because we strip spaces when copying the - extension into the structure. Therefore, use the copied item to - place the item into the list. (found by lmadsen on -dev, fixed by - me) - -2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell@digium.com> - - * main/astobj2.c, include/asterisk/astobj2.h: When compiling with - DEBUG_THREADS, report the real file/func/line for - ao2_lock/ao2_unlock - -2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf@digium.com> - - * res/res_features.c: (closes issue #13871) Reported by: mdu113 - This one is totally my fault. The code doesn't even create a - bridge if the channel CDR has POST_DISABLED. I didn't check for - that at the end of the bridge. Fixed with a few small insertions. - Tested. Looks good. No cdr generated, no crash, no unnecc. data - objects created either. - -2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: This change had somehow gotten reverted due to - a completely unrelated commit. Thanks to Theo Belder on the - Asterisk-dev list for pointing this out. - - * include/asterisk/file.h, main/frame.c, main/file.c, - include/asterisk/frame.h: There was an issue when attempting to - reference an embedded frame in a freed ast_filestream. This patch - makes use of the ao2 functions to make sure that we do not free - an ast_filestream structure until the embedded ast_frame has been - "freed" as well. (closes issue #13496) Reported by: fst-onge - Patches: filestream_frame_1_4.diff uploaded by putnopvut (license - 60) Tested by: putnopvut Closes AST-89 - - * channels/chan_sip.c: We don't handle 4XX responses to BYE well. - According to section 15 of RFC 3261, we should terminate a dialog - if we receive a 481 or 408 in response to our BYE. Since I am - aware of at least one phone manufacturer who may sometimes send a - 404 as well, I am being liberal and saying that any 4XX response - to a BYE should result in a terminated dialog. (closes issue - #12994) Reported by: pabelanger Patches: 12994.patch uploaded by - putnopvut (license 60) Closes AST-129 - - * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup - cause on the calling channel in the case that ast_call() fails. - For incoming SIP channels, this was causing us to send a 603 - instead of a 486 when the call-limit was reached on the - destination channel. (closes issue #13867) Reported by: still_nsk - Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: - blitzrage - -2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c: Merged revision 157977 from - https://origsvn.digium.com/svn/asterisk/team/group/issue8824 - ........ Fixes JIRA ABE-1726 The dial extension could be empty if - you are using MISDN_KEYPAD to control ISDN provider features. - -2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming@digium.com> - - * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, - channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels, - main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, - codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, - channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules, - res/snmp, res/Makefile: the gcc optimizer frequently finds broken - code (use of uninitalized variables, unreachable code, etc.), - which is good. however, developers usually compile with the - optimizer turned off, because if they need to debug the resulting - code, optimized code makes that process very difficult. this - means that we get code changes committed that weren't adequately - checked over for these sorts of problems. with this build system - change, if (and only if) --enable-dev-mode was used and - DONT_OPTIMIZE is turned on, when a source file is compiled it - will actually be preprocessed (into a .i or .ii file), then - compiled once with optimization (with the result sent to - /dev/null) and again without optimization (but only if the first - compile succeeded, of course). while making these changes, i did - some cleanup work in Makefile.rules to move commonly-used - combinations of flag variables into their own variables, to make - the file easier to read and maintain - -2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Add some missing invite state changes - necessary in the sip_write function. Not setting the invite state - correctly on the call was resulting in the Record application - leaving empty files. I also have updated the doxygen comment next - to the declaration of the INV_EARLY_MEDIA constant to reflect - that we also use this state when we *send* a 18X response to an - INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: - sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco - (license 162) Tested by: putnopvut - -2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler@digium.com> - - * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel - This fix is the result of a bug fix in ast_app_separate_args - r124395. If an argument does not exist it should always be set to - a null string rather than a null pointer. - -2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, channels/chan_local.c, res/res_features.c, - include/asterisk/channel.h, apps/app_followme.c: Fix a crash in - the end_bridge_callback of app_dial and app_followme which would - occur at the end of an attended transfer. The error occurred - because we initially stored a pointer to an ast_channel which - then was hung up due to a masquerade. This commit adds a "fixup" - callback to the bridge_config structure to allow for - end_bridge_callback_data to be changed in the case that a new - channel pointer is needed for the end_bridge_callback. - -2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, Makefile.rules: when an individual directory dist-clean - is run, run clean in that directory first, and when running - top-level dist-clean, do not run subdirectory clean operations - twice - - * Makefile.moddir_rules: dist-clean should remove dependency - information files as well - - * contrib/asterisk-ng-doxygen: major update to doxygen - configuration file: 1) update to doxygen 1.5.x style file, as - used in trunk 2) tell doxygen where are header files are, so - include-file processing can be done 3) make all macros that are - used to define variables/functions be expanded, so that doxygen - will properly document the resulting variable/function 4) make - all macros that are used to provide the contents of a variable - (structure) be expanded, so that doxygen will be able to document - the resulting fields 5) suppress compiler attributes - (__attribute__(xxx)) from being seen by doxygen, so it will - properly match up function definition and usage (for an example - of th effect of this, look at the doxygen docs for ast_log() from - before and afte this commit) - -2008-11-14 15:18 +0000 [r156816] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: If the prompt to reenter a voicemail - password timed out, it resulted in the password not being saved, - even if the input matched what you gave when first prompted to - enter a new password. This is because the return value of - ast_readstring was checked, but not checked properly. This bug - was discovered by Jared Smith during an Asterisk training course. - Thanks for reporting it! - -2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher <tlesher@digium.com> - - * apps/app_while.c: ast_waitfordigit() requires that the channel be - up, for no good logical reason. This prevents While/EndWhile from - working within the "h" extension. Reported by: jgalarneau (for - ABE C.2) Fixed by: me - - * main/manager.c: Provide more space for all the data which can - appear in an originating channel name. (closes issue #13398) - Reported by: bamby Patches: manager.c.diff uploaded by bamby - (license 430) - -2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming <kpfleming@digium.com> - - * configure, autoconf/ast_gcc_attribute.m4: revert this change... - non-functional changes don't belong here - - * configure, autoconf/ast_gcc_attribute.m4: correct minor syntax - error... no functional change - -2008-11-12 21:18 +0000 [r156386] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c: When using call limits under 1 second, infinite - call lengths are allowed, instead. (closes issue #13851) Reported - by: ruddy - -2008-11-12 19:36 +0000 [r156297] Steve Murphy <murf@digium.com> - - * main/pbx.c: It turns out that the 0x0XX00 codes being returned - for N, X, and Z are off by one, as per conversation with jsmith - on #asterisk-dev; he was teaching a class and disconcerted that - this published rule was not being followed, with patterns _NXX, - _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should - have been. This change, tested on these 3 patterns now picks the - proper one. However, this change may surprise users who set up - dialplans based on previous behavior, which has been there for - what, 2 and half years or so now. - -2008-11-12 19:26 +0000 [r156294] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c: If the SLA thread is not started, then reload - causes a memory leak. (closes issue #13889) Reported by: eliel - Patches: app_meetme.c.patch uploaded by eliel (license 64) - -2008-11-12 19:10 +0000 [r156289] Jeff Peeler <jpeeler@digium.com> - - * apps/app_meetme.c: For whatever reason, gcc only warned me about - the possible use of an uninitialized variable when compiling - 1.6.1. - -2008-11-12 18:39 +0000 [r156229] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Revert revision 132506, since it - occasionally caused IAX2 HANGUP packets not to be sent, and - instead, schedule a task to destroy the iax2 pvt structure 10 - seconds later. This allows the IAX2 HANGUP packet to be queued, - transmitted, and ACKed before the pvt is destroyed. (closes issue - #13645) Reported by: dzajro Patches: - 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14) - Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ - -2008-11-12 17:53 +0000 [r156178] Jeff Peeler <jpeeler@digium.com> - - * apps/app_meetme.c: (closes issue #13173) Reported by: pep This - change adds an announce_thread responsible for playing - announcements to an existing conference. This allows all - announcing to be immediately stopped if necessary but more - importantly allows other threads that need to play something to - not block. There are multiple benefits to this, but the actual - bug is for solving the scenario for a channel to be unusable - after hang up for the entire duration of the parting - announcement. The parting announcement can be extremely long - depending on what the user recorded upon joining the conference. - Reviewed by Russell on Review Board: - http://reviewboard.digium.com/r/25/ - -2008-11-12 17:38 +0000 [r156167] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: When doing some tests, I was having a crash at - the end of every call if an attended transfer occurred during the - call. I traced the cause to the CDR on one of the channels being - NULL. murf suggested a check in the end bridge callback to be - sure the CDR is non-NULL before proceeding, so that's what I'm - adding. - -2008-11-12 17:29 +0000 [r156164] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Move the sanity check that makes sure "always - fork" is not set along with the console option to be after the - code that reads options from asterisk.conf. This resolves a - situation where Asterisk can start taking up 100% when - misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting - me log in to his system to figure out what was causing the 100% - CPU problem.) - -2008-11-10 21:07 +0000 [r155861] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Channel drivers assume that when their - indicate callback is invoked, that the channel on which the - callback was called is locked. This patch corrects an instance in - chan_agent where a channel's indicate callback is called directly - without first locking the channel. This was leading to some - observed locking issues in chan_local, but considering that all - channel drivers operate under the same expectations, the generic - fix in chan_agent is the right way to go. AST-126 - -2008-11-10 20:49 +0000 [r155803] Tilghman Lesher <tlesher@digium.com> - - * doc/valgrind.txt: I got tired of saying this in every single - bugnote referring to this file. - -2008-11-09 01:08 +0000 [r155553] Sean Bright <sean.bright@gmail.com> - - * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h, - apps/app_followme.c: Use static functions here instead of nested - ones. This requires a small change to the ast_bridge_config - struct as well. To understand the reason for this change, see the - following post: - http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html - -2008-11-07 22:27 +0000 [r155398] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Clarify error message. (closes issue #13809) - Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded - by Corydon76 (license 14) Tested by: denke - -2008-11-06 19:45 +0000 [r155011] Mark Michelson <mmichelson@digium.com> - - * configs/voicemail.conf.sample: The documentation listed the - ability to set 'maxmsg' per context. The truth is that you can - only set this in the general section or per mailbox. Thus I am - updating the sample config file to be more accurate. Thanks to - sasargen on IRC for bringing up this issue. - -2008-11-05 16:44 +0000 [r154724] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: The logic of a strcasecmp call was - reversed (closes issue #13841) Reported by: clegall_proformatique - -2008-11-05 16:06 +0000 [r154685] Steve Murphy <murf@digium.com> - - * main/channel.c: This fix was prompted by communication from user, - who was seeing thousands of error logs... looks like EAGAIN. Made - such uninteresting. - -2008-11-04 20:49 +0000 [r154365] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: On busy systems, it's possible for the - values checked within a single line of code to change, unless the - structure is locked to ensure a consistent state. (closes issue - #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt - uploaded by Corydon76 (license 14) Tested by: kowalma - -2008-11-04 19:01 +0000 [r154266] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to - the wrong state when it receives the indication - AST_CONTROL_RINGING. - -2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_h323.c: Make the monitor thread non-detached, so it - can be joined (suggested by Russell on -dev list). - - * apps/app_voicemail.c: Attempting to expunge a mailbox when the - mailstream is NULL will crash Asterisk. (Closes issue #13829) - Reported by: jaroth Patch by: me (modified jaroth's patch) - - * main/rtp.c: Remove the potential for a division by zero error. - (Closes issue #13810) - -2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c, - main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c, - res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in - this branch on the first pass - -2008-11-02 19:51 +0000 [r153651] Russell Bryant <russell@digium.com> - - * include/asterisk/features.h: features.h depends on linkedlists.h, - so include it - -2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming <kpfleming@digium.com> - - * utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c, - funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c, - channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c, - pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c, - pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c, - formats/format_wav.c, apps/app_festival.c, - main/db1-ast/hash/hash_page.c, main/translate.c, - res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c, - utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c, - pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h, - apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c, - utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c, - apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y, - apps/app_authenticate.c, formats/format_wav_gsm.c, - res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of - potential problems found by gcc 4.3.x, primarily bare strings - being passed to printf()-like functions and ignored results from - read()/write() and friends - -2008-10-31 22:36 +0000 [r153270] Terry Wilson <twilson@digium.com> - - * res/res_features.c, apps/app_followme.c: Add end_bridge_callback - for app_follome and add AUTOLOOP flag to res_features - -2008-10-31 Tilghman Lesher <tlesher@digium.com> - - * Asterisk 1.4.23-rc1 released. - -2008-10-31 16:30 +0000 [r153114] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Turn off qualify on uncached realtime peers. - (Closes issue #13383) - -2008-10-31 15:45 +0000 [r153095] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h: - Recent CDR fixes moved execution of the 'h' exten into the - bridging code, so variables that were set after ast_bridge_call - was called would not show up in the 'h' exten. Added a callback - function to handle setting variables, etc. from w/in the bridging - code. Calls back into a nested function within the function - calling ast_bridge_call (closes issue #13793) Reported by: - greenfieldtech - -2008-10-30 20:58 +0000 [r152992] Sean Bright <sean.bright@gmail.com> - - * bootstrap.sh: The -I argument to aclocal needs a space before the - include directory name. - -2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_h323.c: Cannot join detached threads. See - http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html - (Closes issue #13400) - - * channels/chan_local.c: Unlock before returning, when extension - doesn't exist. (closes issue #13807) Reported by: eliel Patches: - chan_local.c.patch uploaded by eliel (license 64) - -2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming <kpfleming@digium.com> - - * main/cdr.c: instead of comparing the string pointer to 0, let's - compare the value that was actually parsed out of the string - (found by sparse) - -2008-10-29 05:23 +0000 [r152539] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes - issue #13795) Reported by: andrew53 Patches: - chan_sip_sizeof.patch uploaded by andrew53 (license 519) - -2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy <murf@digium.com> - - * configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c: - A little documentation cross-ref between features and dial and - queue... I wasted some time (stupidly) trying to get the - one-touch parking stuff working, because it didn't occur to me - that I had to also have the corresponding options in the dial - command! Duh! (In all this time, I never set this up before!) So, - to keep some poor fool from suffering the same fate, I made the - features.conf.sample file mention the corresponding opts in - dial/queue; and the docs for dial/app specifically mention the - corresponding decls in the feature.conf file. I hope this doesn't - spoil some vast, eternal plan... - - * apps/app_dial.c, res/res_features.c, funcs/func_channel.c, - include/asterisk/pbx.h, apps/app_queue.c: The magic trick to - avoid this crash is not to try to find the channel by name in the - list, which is slow and resource consuming, but rather to pay - attention to the result codes from the ast_bridge_call, to which - I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are - returned when a channel is parked. If you get AST_PBX_KEEPALIVE, - then don't touch the channel pointer. If you get - AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then - don't touch the peer pointer. Updated the several places where - the results from a bridge were not being properly obeyed, and - fixed some code I had introduced so that the results of the - bridge were not overridden (in trunk). All the places that - previously tested for AST_PBX_NO_HANGUP_PEER now have to check - for both AST_PBX_NO_HANGUP_PEER and - AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common - parking scenarios: 1. A calls B; B answers; A parks B; B hangs up - while A is getting the parking slot announcement, immediately - after being put on hold. 2. A calls B; B answers; A parks B; B - hangs up after A has been hung up, but before the park times out. - 3. A calls B; B answers; B parks A; A hangs up while B is getting - the parking slot announcement, immediately after being put on - hold. 4. A calls B; B answers; B parks A; A hangs up after B has - been hung up, but before the park times out. No crash. I also ran - the scenarios above against valgrind, and accesses looked good. - -2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Quoting in the wrong direction (Fixes - AST-107) - - * apps/app_dial.c: Reset all DIAL variables back to blank, in case - Dial is called multiple times per call (which could otherwise - lead to inconsistent status reports). (closes issue #13216) - Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded - by Corydon76 (license 14) Tested by: ruddy - -2008-10-27 23:28 +0000 [r152286] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Buffer policy setting for half is not - needed. - -2008-10-27 21:32 +0000 [r152215] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c: Inherit ALL elements of CallerID across a - local channel. (closes issue #13368) Reported by: Peter Schlaile - Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 - (license 14) - -2008-10-26 20:23 +0000 [r152059] Sean Bright <sean.bright@gmail.com> - - * funcs/func_strings.c: Since passing \0 as the second argument to - strchr is valid (and will match the trailing \0 of a string) we - need to check that first, otherwise we end up with incorrect - results. Fix suggested by reporter. (closes issue #13787) - Reported by: meitinger - -2008-10-25 10:59 +0000 [r151905] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Move AMI initialization to occur after loading - modules. This prevents a deadlock when someone tries to initiate - a module reload from the AMI just as Asterisk is starting. - (closes issue #13778) Reported by: hotsblanc Fix suggested by - hotsblanc - -2008-10-23 16:04 +0000 [r151763] Terry Wilson <twilson@digium.com> - - * configs/features.conf.sample, res/res_features.c, CHANGES: - Backport fix from 1.6.0 that allows you to set - parkedcalltransfers=no|caller|callee|both, but default to both - which would be the equivalent of the existing behaviour. The - problem was that if someone parked a call, the callee and caller - would both get assigned the builtin transfer feature, which would - not only be potentially giving someone the ability to transfer - themselves when they shouldn't have it, but would also dissallow - reinviting the media off of the call. (closes issue #12854) - Reported by: davidw Patches: parkingfix4.diff.txt uploaded by - otherwiseguy Tested by: davidw, otherwiseguy - -2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming <kpfleming@digium.com> - - * autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4, - configure.ac: rename this macro to properly reflect what it does - - * autoconf/ast_check_pwlib.m4 (added), autoconf (added), - autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4 - (added), configure, autoconf/ast_gcc_attribute.m4 (added), - bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added), - autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added), - autoconf/ast_c_compile_check.m4 (added), - autoconf/ast_c_define_check.m4 (added), - autoconf/ast_prog_egrep.m4 (added), - autoconf/ast_check_openh323.m4 (added), - autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4 - (added), acinclude.m4 (removed): break up acinclude.m4 into - individual files, which will make it easier to maintain, easier - to add new macros (less patching) and will ease maintenance of - these macros across Asterisk branches - -2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke <bweschke@btwtech.com> - - * main/asterisk.c: As per kpfleming's comments to the prior commit, - I'm reverting some of the changes here. A comment was made in bug - #13726 "3. The same mistake as in (2) is done in a few other - places in the code that check for: #if defined(HAVE_ZAPTEL) || - defined(HAVE_DAHDI) Harmless, but still incorrect." In the case - of main/asterisk.c, this is not incorrect because without - HAVE_ZAPTEL defined, we're missing the include for ioctl and the - namespace that defines DAHDI_TIMERCONFIG which is still required - when using Zaptel with the 1.4 branch. - - * main/asterisk.c: Fix the 1.4 branch compile again broken with - r150557 when using with Zaptel and not DAHDI (closes issue - #13740) reported by: jmls patch by: bweschke - -2008-10-18 01:42 +0000 [r150816] BJ Weschke <bweschke@btwtech.com> - - * main/manager.c: Using the GetVar handler in AMI is potentially - dangerous (insta-crash [tm]) when you use a dialplan function - that requires a channel and then you don't provide one or provide - an invalid one in the Channel: parameter. We'll handle this - situation exactly the same way it was handled in pbx.c back on - r61766. We'll create a bogus channel for the function call and - destroy it when we're done. If we have trouble allocating the - bogus channel then we're not going to try executing the function - call at all and run the risk of crashing. (closes issue #13715) - reported by: makoto patch by: bweschke - -2008-10-17 17:18 +0000 [r150637] Steve Murphy <murf@digium.com> - - * res/res_features.c: Interesting crash. In this case, you exit the - bridge with peer completely GONE. I moved the channel find call - up to cover the whole peer CDR reset code segment. This appears - to solve the crash without changing the logic at all. - -2008-10-17 15:31 +0000 [r150557] Jason Parker <jparker@digium.com> - - * main/asterisk.c, main/channel.c, channels/chan_dahdi.c, - configure, configure.ac: Correctly allow chan_dahdi to compile - against older versions of Zaptel. Don't always define - HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor - cleanup to make things clear. (closes issue #13726) Reported by: - tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46) - -2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson <mmichelson@digium.com> - - * main/manager.c: Reverting changes from commits 150298 and 150301 - since I was mistakenly under the assumption that dialplan - functions *always* required that a channel be present. I need to - go home earlier, I think :) - - * main/manager.c: And don't forget to return on the error condition - - * main/manager.c: Don't try to call a dialplan function's read - callback from the manager's GetVar handler if an invalid channel - has been specified. Several dialplan functions, including CHANNEL - and SIP_HEADER, do not check for NULL-ness of the channel being - passed in. (closes issue #13715) Reported by: makoto - -2008-10-16 15:56 +0000 [r150124] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c: Fix memory leak found by customer - -2008-10-16 15:26 +0000 [r150056] Steve Murphy <murf@digium.com> - - * cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and - RYM-150398 and AST-103 in internal Digium bug trackers. These - fixes address a really subtle memory corruption problem that - would happen in machines heavily loaded in production - environments. The corruption would always take the form of the - STMT object getting nulled out and one of the unixODBC calls - would crash trying to access statement->connection. It isn't - fully proven yet, but the server has now been running 2.5 days - without appreciable memory growth, or any gain of %cpu, and no - crashes. Whether this is the problem or not on that server, these - fixes are still warranted. As it turns out, **I** introduced - these errors unwittingly, when I corrected another crash earlier. - I had formed the build_query routine, and failed to remove - mutex_unlock calls in 3 places in the transplanted code. These - unlocks would only happen in error situations, but unlocking the - mutex early set the code up for a catastrophic failure, it - appears. It would happen only once every 100K-200K or more calls, - under heavy load... but that is enough. If another crash occurs, - with the same MO, I'll come back and remove my confession from - the log, and we'll keep searching, but the fact that we have - Asterisk dying from an asynchronous wiping of the STMT object, - only on some connection error, and that the server has lived for - 2.5 days on this code without a crash, sure make it look like - this was the problem! Also, in several points, Statement handles - are set to NULL after SQLFreeHandle. This was mainly for - insurance, to guarantee a crash. As it turns out, the code does - not appear to be attempting to use these freed pointers. Asterisk - owes a debt of gratitude to Federico Alves and Frediano Ziglio - for their untiring efforts in finding this bug, among others. - -2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke <bweschke@btwtech.com> - - * CHANGES: Another documentation fix. (closes issue #13708) - - * configs/agents.conf.sample: An update to the - documentation/example of agents.conf.sample with the correct - parameter for this feature as defined in chan_agent.c (closes - issue #13709) - -2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix some problems when parsing SIP messages - that have the maximum number of headers or body lines that we - support - -2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Change this warning to an error message. - Suggestion comes from Sean Bright. Thanks Sean! - - * channels/chan_sip.c: Call register_peer_exten even in the case - that the peer's IP/port does not change. (closes issue #13309) - Reported by: dimas Patches: v2-13309.patch uploaded by dimas - (license 88) - - * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance - period for sync-triggered audiohooks so that if packetization of - audio is close (but not equal) we don't end up flushing the - audiohooks over small inconsistencies in synchronization. Related - to issue #13005, and solves the issue for most people who were - experiencing the problem. However, a small number of people are - still experiencing the problem on long calls, so I am not closing - the issue yet - - * apps/app_queue.c: Update the queue with the correct number of - calls and whether the call was completed within the service level - when a transfer takes place. This way, we do not "break" the - leastrecent and fewestcalls strategies by not logging a call - until after the transferred call has ended. (closes issue #13395) - Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded - by Marquis (license 32) - - * channels/chan_sip.c: Don't allow reserved characters to be used - in register lines in sip.conf. (closes issue #13570) Reported by: - putnopvut - -2008-10-14 20:09 +0000 [r149061] Tilghman Lesher <tlesher@digium.com> - - * apps/app_waitforsilence.c: Check correct values in the return of - ast_waitfor(); also, get rid of a possible memory leak. (closes - issue #13658) Reported by: explidous Patch by: me - -2008-10-14 19:05 +0000 [r148990] Leif Madsen <lmadsen@digium.com> - - * CHANGES: Add in some missing updates to the CHANGES file for - sip.conf (closes issue #13100) Reported and patch by: - gknispel_proformatique - -2008-10-14 19:03 +0000 [r148916-148987] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Some compilers warn, some don't. Fixing. - - * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean, - even when UTF-8 characters are used in headers like 'Subject' and - 'To'. Closes AST-107. - -2008-10-14 17:33 +0000 [r148912] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c: Deadlock prevention in chan_local. (closes - issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded - by putnopvut (license 60) Tested by: tacvbo - -2008-10-14 10:30 +0000 [r148611-148736] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: on Ubuntu (at least), recent versions of ld in binutils - delete all debugging symbols when -x is supplied; since the - reasons why -x is being passed are lost in the mists of time, - remove it so debugging will work properly - - * main/translate.c: it would be nice if this message printing code - had actually been tested before it was committed... - -2008-10-10 16:25 +0000 [r147997-148257] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: User not notified of temporary greeting, if - ODBC storage is in use. (closes issue #13659) Reported by: - moliveras Patches: 20081009__bug13659.diff.txt uploaded by - Corydon76 (license 14) Tested by: moliveras - - * apps/app_voicemail.c: When blank, callerid name and number should - display "unknown caller" in voicemail emails. (Closes issue - #13643) - -2008-10-09 18:56 +0000 [r147941] Jeff Peeler <jpeeler@digium.com> - - * res/res_features.c: (closes issue #13139) Reported by: krisk84 - Tested by: krisk84 This change prevents a call that is placed in - the parkinglot to be picked up before the PBX is finished. If - another extension dials the parking extension before the PBX - thread has completed at minimum warnings will occur about the PBX - not properly being terminated. At worst, a crash could occur. - -2008-10-08 22:22 +0000 [r147681] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: when parsing a text configuration option, - ensure that the buffer on the stack is actually large enough to - hold the legal values of that option, and also ensure that - sscanf() knows to stop parsing if it would overrun the buffer - (without these changes, specifying "buffers=...,immediate" would - overflow the buffer on the stack, and could not have worked as - expected) - -2008-10-08 14:51 +0000 [r147517] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: If we receive DTMF make sure that the - state of the speech structure goes back to being not ready. - (issue #LUMENVOX-8) - -2008-10-07 23:14 +0000 [r147429-147430] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: revert this change until i can understand - why it results in locking order changes - - * channels/chan_dahdi.c: don't start a PBX on incoming PRI call - channels until after we're done setting channel variables and - other things on the channel, otherwise the channel might go away - (if the dialplan hangs up quickly) before we are done, which - results in a spectacular crash - -2008-10-07 16:48 +0000 [r147193] Sean Bright <sean.bright@gmail.com> - - * apps/app_voicemail.c: Make 'imapsecret' an alias to - 'imappassword' in voicemail.conf. - -2008-10-06 20:52 +0000 [r146711-146799] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_callerid.c, apps/app_speech_utils.c, - funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, - channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c, - funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c: - Dialplan functions should not actually return 0, unless they have - modified the workspace. To signal an error (and no change to the - workspace), -1 should be returned instead. (closes issue #13340) - Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt - uploaded by Corydon76 (license 14) - - * channels/chan_local.c: Check whether an extension exists in the - _call method, rather than the _alloc method, because we need to - evaluate the callerid (since that data affects whether an - extension exists). (closes issue #13343) Reported by: efutch - Patches: 20080915__bug13343.diff.txt uploaded by Corydon76 - (license 14) Tested by: efutch - -2008-10-06 15:57 +0000 [r146643] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: ensure that the private structure for - pseudo channels is created without 'leaking' configuration data - from other configured channels (closes issue #13555) Reported by: - jeffg Patches: issue_13555.patch uploaded by kpfleming (license - 421) Tested by: jeffg - -2008-10-05 21:17 +0000 [r146448] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Fix silly formatting. - -2008-10-03 22:51 +0000 [r146244] Sean Bright <sean.bright@gmail.com> - - * apps/app_rpt.c: Change some preprocessor macros to struct - definitions so that we get app_rpt to build with DAHDI. (closes - issue #13576) Reported by: blitzrage - -2008-10-03 20:44 +0000 [r146129] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/features.h, res/res_features.c, res/res_agi.c: - (closes issue #13425) Reported by: mdu113 Tested by: mdu113 - Similar to r143204, masquerade the channel in the case of Park - being called from AGI. - -2008-10-03 17:12 +0000 [r146026] Steve Murphy <murf@digium.com> - - * res/res_features.c: (closes issue #13579) Reported by: dwagner - (closes issue #13584) Reported by: dwagner Tested by: murf, - putnopvut The thought occurred to me that the res= from the - extension spawn was ending up being returned from the bridge. - "Thou shalt not poison the return value". Made the change and it - appears to allow blind xfers to work as normal. If I'm wrong, - reopen the bugs. But it looks good to me! Many thanks to - putnopvut for helping me reproduce this! - -2008-10-02 16:39 +0000 [r145751-145839] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_odbc.c: Backport support for some of the keyword - modifications used in 1.6 (while warning that some options aren't - really supported) and add some warning messages. Some credit to - oej, who was complaining in #asterisk-dev. - - * res/res_odbc.c: Some sanity checks that may have led to prior - crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup - of incorrectly-used constants. - -2008-10-01 17:18 +0000 [r145479] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql - script to create the setinterfacevar column. (closes issue - #13549) Reported by: fiddur - -2008-10-01 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.22 released. - -2008-09-09 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.22-rc5 released. - -2008-09-09 15:40 +0000 [r142063] Russell Bryant <russell@digium.com> - - * res/res_features.c: Ensure that the stored CDR reference is still - valid after the bridge before poking at it. Also, keep the - channel locked while messing with this CDR. (fixes crashes - reported in issue #13409) - -2008-09-08 21:10 +0000 [r141809] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix pedantic mode of chan_sip to only check - the remote tag of an endpoint once a dialog has been confirmed. - Up until that point, it is possible and legal for the far-end to - send provisional responses with a different To: tag each time. - With this patch applied, these provisional messages will not - cause a matching problem. (closes issue #11536) Reported by: ibc - Patches: 11536v2.patch uploaded by putnopvut (license 60) - -2008-09-08 21:02 +0000 [r141806] Russell Bryant <russell@digium.com> - - * main/pbx.c: When doing an async goto, detect if the channel is - already in the middle of a masquerade. This can happen when - chan_local is trying to optimize itself out. If this happens, - fail the async goto instead of bursting into flames. (closes - issue #13435) Reported by: geoff2010 - -2008-09-08 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.22-rc4 released. - -2008-09-08 20:15 +0000 [r141741] Jason Parker <jparker@digium.com> - - * Makefile, redhat (removed): Remove RPM package targets from - Makefile (and all associated parts). This has never worked in - 1.4, and we decided that it makes no sense to be done here. There - are many distros out there that already have "proper" spec files - that can be (re)used. Closes issue #13113 Closes issue #10950 - Closes issue #10952 - -2008-09-08 16:26 +0000 [r141678] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Actually use Zaptel CFLAGS if using - Zaptel instead of DAHDI This fixes building against Zaptel when - using a custom path - -2008-09-06 20:13 +0000 [r141565] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: This fix comes from Joshua Colp The - Brilliant, who, given the trace, came up with a solution. This - will most likely will close 13235 and 13409. I'll wait till - Monday to verify, and then close these bugs. - -2008-09-06 15:23 +0000 [r141503] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Reverting behavior change (AGI should not exit - non-zero on SUCCESS) (closes issue #13434) Reported by: - francesco_r - -2008-09-05 21:10 +0000 [r141217-141366] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Agent's should not try to call a channel's - indicate callback if the channel has been hung up. It will likely - crash otherwise ABE-1159 - - * apps/app_voicemail.c: Since greetings are not stored in IMAP, we - should not be DISPOSE'ing of them the same way we do with other - messages. (closes issue #13414) Reported by: mthomasslo Patches: - 13414v2.patch uploaded by putnopvut (license 60) Tested by: - mthomasslo - - * channels/chan_sip.c: Commit 140417 had a logic flaw in it which - caused port 5060 to always be used when dialing a peer if no - explicit port was specified. This broke the behavior of - implicitly using the port from which the peer registered if no - port is specified. This commit fixes the logic flaw. (closes - issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded - by putnopvut (license 60) Tested by: mdu113 - -2008-09-05 14:15 +0000 [r141094-141156] Steve Murphy <murf@digium.com> - - * main/channel.c: A small change to prevent double-posting of - CDR's; thanks to Daniel Ferrer for bringing it to our attention - - * pbx/ael/ael-test/ref.ael-vtest25 (added), - pbx/ael/ael-test/ael-vtest25/extensions.ael (added), - pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c, - pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue - #13357) Reported by: pj Tested by: murf (closes issue #13416) - Reported by: yarns Tested by: murf If you find this message - overly verbose, relax, it's probably not meant for you. This - message is meant for probably only two people in the whole world: - me, or the poor schnook that has to maintain this code because - I'm either dead or unavailable at the moment. This fix solves two - reports, both having to do with embedding a function call in a - ${} construct. It was tricky because the funccall syntax has - parenthesis () in it. And up till now, the 'word' token in the - flex stuff didn't allow that, because it would tend to steal the - LP and RP tokens. To be truthful, the "word" token was the - trickiest, most unstable thing in the whole lexer. I was lucky it - made this long without complaints. I had to choose every - character in the pattern with extreme care, and I knew that - someday I'd have to revisit it. Well, the day has come. So, my - brilliant idea (and I'm being modest), was to use the surrounding - ${} construct to make a state machine and capture everything in - it, no matter what it contains. But, I have to now treat the word - token like I did with comments, in that I turn the whole thing - into a state-machine sort of spec, with new contexts - "curlystate", "wordstate", and "brackstate". Wait a minute, - "brackstate"? Yes, well, it didn't take very many regression - tests to point out if I do this for ${} constructs, I also have - to do it with the $[] constructs, too. I had to create a separate - pcbstack2 and pcbstack3 because these constructs can occur inside - macro argument lists, and when we have two state machines - operating on the same structures we'd get problems otherwise. I - guess I could have stopped at pcbstack2 and had the brackstate - stuff share it, but it doesn't hurt to be safe. So, the pcbpush - and pcbpop routines also now have versions for "2" and "3". I had - to add the {KEYWORD} construct to the initial pattern for "word", - because previously word would match stuff like "default7", - because it was a longer match than the keyword "default". But, - not any more, because the word pattern only matches only one or - two characters now, and it will always lose. So, I made it the - winner again by making an optional match on any of the keywords - before it's normal pattern. I added another regression test to - make sure we don't lose this in future edits, and had to fix just - one regression, where it no longer reports a 'cascaded' error, - which I guess is a plus. I've given some thought as to whether to - apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I - decided to put it in 1.4 because one of the bug reports was - against 1.4; and it is unexpected that AEL cannot handle this - situation. It actually reduced the amount of useless "cascade" - error messages that appeared in the regressions (by one line, - ehhem). There is a possible side-effect in that it does now do - more careful checking of what's in those ${} constructs, as far - as matching parens, and brackets are concerned. Some users may - find a an insidious problem and correct it this way. This should - be exceedingly rare, I hope. - -2008-09-04 17:00 +0000 [r141028] Jeff Peeler <jpeeler@digium.com> - - * res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes - multiple parking problems: Crash when executing a park on an - extension dialed by AGI due to not returning the proper return - code. Crash when using a builtin feature that was a subset of a - enabled dynamic feature. Crash due to always hanging up the peer - despite the fact that the peer was supposed to be parked. - -2008-09-03 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.22-rc3 released. - -2008-09-03 14:29 +0000 [r140850] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix voicemail forwarding when using ODBC - storage. (closes issue #13387) Reported by: moliveras Patches: - 13387.patch uploaded by putnopvut (license 60) Tested by: - putnopvut, moliveras - -2008-09-03 13:24 +0000 [r140816] Russell Bryant <russell@digium.com> - - * main/poll.c: Don't freak out if the poll emulation receives NULL - for the pollfds array (closes issue #13307) Reported by: jcovert - -2008-09-02 23:47 +0000 [r140751] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: After adding the context checking to - app_voicemail for IMAP storage, I left out a crucial place to - copy the context to the vm_state structure. This is the - correction. - -2008-09-02 23:36 +0000 [r140670-140747] Steve Murphy <murf@digium.com> - - * main/cdr.c: I am turning the warnings generated in ast_cdr_free - and post_cdr into verbose level 2 messages. Really, they matter - little to end users. You either get the CDR's you wanted, or you - don't, and it is a bug. - - * main/channel.c: After reconsidering, with respect to 13409, - ast_cdr_detach should be OK, better in fact, than ast_cdr_free, - which generates lots of useless warnings that will undoubtably - generate complaints. - - * main/channel.c, main/pbx.c: (closes issue #13409) Reported by: - tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by - tomaso (license 564) I basically spent the day, verifying that - this patch solves the problem, and doesn't hurt in non-problem - cases. Why valgrind did not plainly reveal this leak absolutely - mystifies and stuns me. Many, many thanks to tomaso for finding - and providing the fix. - -2008-09-02 18:14 +0000 [r140605] Sean Bright <sean.bright@gmail.com> - - * channels/chan_iax2.c: Make sure to use the correct length of the - mohinterpret and mohsuggest buffers when copying configuration - values. (closes issue #13336) Reported by: - decryptus_proformatique Patches: - chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded - by decryptus (license 555) - -2008-08-29 17:34 +0000 [r140417-140488] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After - working on the ao2_containers branch, I noticed something a bit - strange. In all cases where we provide a callback function to - ao2_container_alloc, the callback function would only return 0 or - CMP_MATCH. After inspecting the ao2_callback() code carefully, I - found that if you're only looking for one specific item, then you - should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will - continue traversing the current bucket until the end searching - for more matches. In cases like chan_iax2 where in 1.4, all the - peers are shoved into a single bucket, this makes for potentially - terrible performance since the entire bucket will be traversed - even if the peer is one of the first ones come across in the - bucket. All the changes I have made were for cases where the - callback function defined was passed to ao2_container_alloc so - that calls to ao2_find could find a unique instance of whatever - object was being stored in the container. - - * apps/app_voicemail.c: Add context checking when retrieving a - vm_state. This was causing a problem for people who had - identically named mailboxes in separate voicemail contexts. This - commit affects IMAP storage only. (closes issue #13194) Reported - by: moliveras Patches: 13194.patch uploaded by putnopvut (license - 60) Tested by: putnopvut, moliveras - - * channels/chan_sip.c: Fix SIP's parsing so that if a port is - specified in a string to Dial(), it is not ignored. (closes issue - #13355) Reported by: acunningham Patches: 13355v2.patch uploaded - by putnopvut (license 60) Tested by: acunningham - -2008-08-27 19:49 +0000 [r140299] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked - when in pedantic mode. The problem was that the wrong tags would - be compared depending on the direction of the call. (closes issue - #13353) Reported by: flefoll Patches: - chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll - (license 244) - -2008-08-26 16:49 +0000 [r140115] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: add HAVE_PRI if define around - dahdi_close_pri_fd - -2008-08-26 16:07 +0000 [r140060] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip. - This code used the return value of a completely unrelated - function to determine whether the scheduler should be run or not. - This would have caused the scheduler to not run in cases where it - should have. Also, leave a note about another scheduler issue - that needs to be addressed at some point. - -2008-08-26 15:57 +0000 [r140056] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir - Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested - by: tzafrir, jpeeler This patch fixes closing open file - descriptors in the case of an error. - -2008-08-26 15:27 +0000 [r140051] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a race condition with the IAX scheduler - thread. A lock and condition are used here to allow newly - scheduled tasks to wake up the scheduler just in case the new - task needs to run sooner than the current wakeup time when the - thread is sleeping. However, there was a race condition such that - a newly scheduled task would not properly wake up the scheduler - or affect the wake up period. The order of execution would have - been: 1) Scheduler thread determines wake up time of N ms. 2) - Another thread schedules a task and signals the condition, with - an execution time of < N ms. 3) Scheduler thread locks and goes - to sleep for N ms. By moving the sleep time determination to - inside the critical section, this possibility is avoided. - -2008-08-26 15:22 +0000 [r140050] Terry Wilson <twilson@digium.com> - - * Makefile: sounds/Makefile installs sounds using the "new" - language directory structure, but languageprefix needs to be set - = yes for sounds in subdirectories (digits/1, etc.) to play as - the correct language. Fix the generation of asterisk.conf to - include languageprefix=yes - -2008-08-26 14:09 +0000 [r140029] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: correct a file location in an error - message - -2008-08-25 21:47 +0000 [r139927] Jeff Peeler <jpeeler@digium.com> - - * main/manager.c: Fix a typo I made. Lesson learned, apply the - patch if one exists. - -2008-08-25 21:31 +0000 [r139909] Sean Bright <sean.bright@gmail.com> - - * build_tools/get_moduleinfo, build_tools/get_makeopts: Some - versions of awk (nawk, for example) don't like empty regular - expressions so be slightly more verbose. (closes issue #13374) - Reported by: dougm Patches: 13374.diff uploaded by seanbright - (license 71) Tested by: dougm - -2008-08-25 20:46 +0000 [r139869] Terry Wilson <twilson@digium.com> - - * channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely - -2008-08-25 15:52 +0000 [r139769] Mark Michelson <mmichelson@digium.com> - - * main/config.c: Fix the logic in config_text_file_save so that if - an UpdateConfig manager action is issued and the file specified - in DstFileName does not yet exist, an error is not returned. - (closes issue #13341) Reported by: vadim Patches: 13341.patch - uploaded by putnopvut (license 60) (with small modification from - seanbright) - -2008-08-25 15:33 +0000 [r139764] Steve Murphy <murf@digium.com> - - * main/pbx.c, res/res_features.c: This patch reverts the changes - made via 139347, and 139635, as users are seeing adverse - difference. I will un-close 13251. Back to the drawing board/ - concept/ beginning/ whatever! - -2008-08-22 22:24 +0000 [r139635] Steve Murphy <murf@digium.com> - - * res/res_features.c: I found some problems with the code I - committed earlier, when I merged them into trunk, so I'm coming - back to clean up. And, in the process, I found an error in the - code I added to trunk and 1.6.x, that I'll fix using this patch - also. - -2008-08-22 21:36 +0000 [r139621] Jeff Peeler <jpeeler@digium.com> - - * main/manager.c: (closes issue #13359) Reported by: Laureano - Patches: originate_channel_check.patch uploaded by Laureano - (license 265) - -2008-08-22 19:45 +0000 [r139456-139553] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/threadstorage.h: Fix compilation when - DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported - by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy - (license 35) - - * main/frame.c: Remove show_frame_stats_deprecated since it is not - used anywhere and causes build errors if building under dev-mode - with TRACE_FRAMES selected in menuselect. (closes issue #13362) - Reported by: snuffy - - * channels/chan_iax2.c: Fix the build. Thanks, mvanbaak! - - * channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting - from incorrect locking order between iax2_pvt and ast_channel - structures. AST-13 - -2008-08-21 23:39 +0000 [r139387] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Fixes loop that could possibly never exit - in the event of a channel never being able to be opened or - specify after a restart. (closes issue #11017) - -2008-08-21 23:03 +0000 [r139347] Steve Murphy <murf@digium.com> - - * main/pbx.c, res/res_features.c: (closes issue #13251) Reported - by: sergee Tested by: murf THis is a bold move for a static - release fix, but I wouldn't have made it if I didn't feel - confident (at least a *bit* confident) that it wouldn't mess - everyone up. The reasoning goes something like this: 1. We simply - cannot do anything with CDR's at the current point (in pbx.c, - after the __ast_pbx_run loop). It's way too late to have any - affect on the CDRs. The CDR is already posted and gone, and the - remnants have been cleared. 2. I was very much afraid that moving - the running of the 'h' extension down into the bridge code (where - it would be now practical to do it), would result in a lot more - calls to the 'h' exten, so I implemented it as another exten - under another name, but found, to my pleasant surprise, that - there was a 1:1 correspondence to the running of the 'h' exten in - the pbx_run loop, and the new spot at the end of the bridge. So, - I ifdef'd out the current 'h' loop, and moved it into the bridge - code. The only difference I can see is the stuff about the - AST_PBX_KEEPALIVE, and hopefully, if this is still an important - decision point, I can replicate it if there are complaints. To be - perfectly honest, the KEEPALIVE situation is not totally clear to - me, and how it relates to a post-bridge situation is less clear. - I suspect the users will point out everything in total clarity if - this steps on anyone's toes! 3. I temporarily swap the bridge_cdr - into the channel before running the 'h' exten, which makes it - possible for users to edit the cdr before it goes out the door. - And, of course, with the endbeforehexten config var set, the - users can also get at the billsec/duration vals. After the h - exten finishes, the cdr is swapped back and processing continues - as normal. Please, all who deal with CDR's, please test this - version of Asterisk, and file bug reports as appropriate! - -2008-08-21 10:11 +0000 [r139283] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4, - too. - -2008-08-20 22:14 +0000 [r139213] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c: Fix a crash in the ChanSpy application. The - issue here is that if you call ChanSpy and specify a spy group, - and sit in the application long enough looping through the - channel list, you will eventually run out of stack space and the - application with exit with a seg fault. The backtrace was always - inside of a harmless snprintf() call, so it was tricky to track - down. However, it turned out that the call to snprintf() was just - the biggest stack consumer in this code path, so it would always - be the first one to hit the boundary. (closes issue #13338) - Reported by: ruddy - -2008-08-20 19:52 +0000 [r139151] Shaun Ruffell <sruffell@digium.com> - - * codecs/codec_dahdi.c: Fix bug where the samples were not accurate - when in G723 mode, which would cause the timestamp field of the - RTP header to be invalid. - -2008-08-20 19:35 +0000 [r139145] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Backport support - for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not - doing so just makes it difficult for people with channels that - are in alarm when Asterisk starts up to get them going once the - alarm is cleared (closes issue #12160) Reported by: tzafrir - Patches: asterisk-chanalarms_14.patch uploaded by tzafrir - (license 46) Tested by: tzafrir - -2008-08-20 17:14 +0000 [r139074] Steve Murphy <murf@digium.com> - - * main/cdr.c: (closes issue #13263) Reported by: brainy Tested by: - murf The specialized reset routine is tromping on the flags field - of the CDR. I made a change to not reset the DISABLED bit. This - should get rid of this problem. - -2008-08-20 15:37 +0000 [r139015] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: sip_read should properly handle a NULL - return from sip_rtp_read. (closes issue #13257) Reported by: - travishein - -2008-08-19 23:22 +0000 [r138949] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL - compatability define for Zaptel - -2008-08-19 23:17 +0000 [r138942] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Reset agent_pvt variables back to the - values in agents.conf (from what the corresponding channel - variables were set to) when the agent logs out. (closes issue - #13098) Reported by: davidw Patches: - 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by - bbryant (license 36) Tested by: davidw - -2008-08-19 22:56 +0000 [r138938] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Add configuration option to - chan_dahdi.conf to allow buffering policy and number of buffers - to be configured per channel. Syntax: buffers=<num of - buffers>,<policy> Where the number of buffers is some - non-negative integer and the policy is either "full", "half", or - "immediate". - -2008-08-19 18:50 +0000 [r138685-138886] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Add a lock and unlock prior to the - destruction of the chanspy_ds lock to ensure that no other - threads still have it locked. While this should not happen under - normal circumstances, it appears that if the spyer and spyee hang - up at nearly the same time, the following may occur. 1. - ast_channel_free is called on the spyee's channel. 2. The chanspy - datastore is removed from the spyee's channel in - ast_channel_free. 3. In the spyer's thread, the spyer attempts to - remove and destroy the datastore from the spyee channel, but the - datastore has already been removed in step 2, so the spyer - continues in the code. 4. The spyee's thread continues and calls - the datastore's destroy callback, chanspy_ds_destroy. This - involves locking the chanspy_ds. 5. Now the spyer attempts to - destroy the chanspy_ds lock. The problem is that in step 4, the - spyee has locked this lock, meaning that the spyer is attempting - to destroy a lock which is currently locked by another thread. - The backtrace provided in issue #12969 supports the idea that - this is possible (and has even occurred). This commit does not - close the issue, but should help in preventing one type of crash - associated with the use of app_chanspy. - - * apps/app_queue.c: Change the inequalities used in app_queue with - regards to timeouts from being strict to non-strict for more - accuracy. (closes issue #13239) Reported by: atis Patches: - app_queue_timeouts_v2.patch uploaded by atis (license 242) - -2008-08-18 16:57 +0000 [r138663] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_dahdi.c: look for transcoder in proper place based - on build against Zaptel or DAHDI - -2008-08-18 11:57 +0000 [r138569] Sean Bright <sean.bright@gmail.com> - - * channels/chan_dahdi.c: You know what's awesome? Code that - compiles... ;) - -2008-08-18 02:05 +0000 [r138516] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: fix compilation warnings - -2008-08-16 01:12 +0000 [r138309-138360] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: fixes use count to properly decrement if - an active dahdi channel is destroyed allowing module to be - unloaded - - * channels/chan_dahdi.c: add forgotten locks around ss_thread_count - in ss_thread for dahdi restart - -2008-08-15 22:33 +0000 [r138258] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: More fixes for - realtime peers. (closes issue #12921) Reported by: Nuitari - Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 - (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 - (license 14) Tested by: Corydon76 - -2008-08-15 21:28 +0000 [r138119-138238] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: initialize condition variable - ss_thread_complete using ast_cond_init - - * channels/chan_dahdi.c: declared static mutexes using - AST_MUTEX_DEFINE_STATIC macro - - * channels/chan_dahdi.c: Fixes the dahdi restart functionality. - Dahdi restart allows one to restart all DAHDI channels, even if - they are currently in use. This is different from unloading and - then loading the module since unloading requires the use count to - be zero. Reloading the module is different in that the signalling - is not changed from what it was originally configured. Also, this - fixes not closing all the file descriptors for D-channels upon - module unload (which would prevent loading the module - afterwards). (closes issue #11017) - -2008-08-15 15:07 +0000 [r138027] Russell Bryant <russell@digium.com> - - * main/autoservice.c: Ensure that when a hangup occurs in - autoservice, that a hangup frame gets properly deferred to be - read from the channel owner when it gets taken out of - autoservice. (closes issue #12874) Reported by: dimas Patches: - v1-12874.patch uploaded by dimas (license 88) - -2008-08-15 14:51 +0000 [r137847-138023] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_strings.c: Additional check for more string specifiers - than arguments. (closes issue #13299) Reported by: adomjan - Patches: 20080813__bug13299.diff.txt uploaded by Corydon76 - (license 14) func_strings.c-sprintf.patch uploaded by adomjan - (license 487) Tested by: adomjan - - * channels/chan_dahdi.c: Oops, wrong direction - - * channels/chan_dahdi.c: When creating the secondary subchannel - name, it is necessary to compare to the existing channel name - without the "Zap/" or "DAHDI/" prefix, since our test string is - also without that prefix. (closes issue #13027) Reported by: - dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer - (license 525) (Slightly modified by me, to compensate for both - names) - -2008-08-14 14:05 +0000 [r137731] Russell Bryant <russell@digium.com> - - * configs/sip.conf.sample: Comments in this config file were - aligned only if your tab size was set to 8. So, convert tabs to - spaces so that things should be aligned regardless of what tab - size you use in your editor. - -2008-08-14 02:03 +0000 [r137677-137679] Kevin P. Fleming <kpfleming@digium.com> - - * Zaptel-to-DAHDI.txt: forgot one module name that changed - - * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c, - build_tools/menuselect-deps.in, configure, configure.ac, - codecs/codec_dahdi.c: add support for Zaptel versions that - contain the new transcoder interface - -2008-08-13 21:35 +0000 [r137580] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Register DAHDISendKeypadFacility - application if dahdi_chan_mode is set to DAHDI + Zap. Mark - ZapSendKeypadFacility application as deprecated on usage. - -2008-08-13 20:46 +0000 [r137527-137530] Kevin P. Fleming <kpfleming@digium.com> - - * Zaptel-to-DAHDI.txt (added): add document describing what users - will need to be aware of when upgrading to this version and using - DAHDI - - * apps/app_meetme.c: remove some more chan_zap references - - * doc/asterisk-conf.txt, channels/chan_dahdi.c: document - dahdichanname option in doc/asterisk-conf.txt make chan_dahdi - read its configuration from zapata.conf if dahdichanname has been - set to 'no' - -2008-08-13 14:33 +0000 [r137348-137405] Sean Bright <sean.bright@gmail.com> - - * doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds - - * cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends - and have it try to store CDR(userfield) if it is set. The new - behavior is to check for the userfield column on module load, and - if it exists, we will store CDR(userfield) when CDRs are written. - A similar patch already went into trunk and 1.6.0. (closes issue - #13290) Reported by: falves11 - -2008-08-11 13:33 +0000 [r137188] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_meetme.c: convert this module to be able to handle DAHDI - or Zaptel (reported on asterisk-users, don't know how this got - missed before) - -2008-08-11 00:20 +0000 [r137138] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c: Deallocate database connection handle on - disconnect, as we allocate another one on connect. (closes issue - #13271) Reported by: dveiga - -2008-08-09 17:11 +0000 [r136999] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate - to 0 if not found instead of empty. pointed out by tzafrir on - #asterisk-dev - -2008-08-09 15:25 +0000 [r136946] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged - revisions 136945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) - | 2 lines Regression fixes for Solaris ........ - -2008-08-08 00:15 +0000 [r136726] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c, - include/asterisk/ael_structs.h: (closes issue #13236) Reported - by: korihor Wow, this one was a challenge! I regrouped and ran a - new strategy for setting the ~~MACRO~~ value; I set it once per - extension, up near the top. It is only set if there is a switch - in the extension. So, I had to put in a chunk of code to detect a - switch in the pval tree. I moved the code to insert the set of - ~~exten~~ up to the beginning of the gen_prios routine, instead - of down in the switch code. I learned that I have to push the - detection of the switches down into the code, so everywhere I - create a new exten in gen_prios, I make sure to pass onto it the - values of the mother_exten first, and the exten next. I had to - add a couple fields to the exten struct to accomplish this, in - the ael_structs.h file. The checked field makes it so we don't - repeat the switch search if it's been done. I also updated the - regressions. - -2008-08-07 18:25 +0000 [r136560] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/menuselect-deps.in, configure, configure.ac: change - the required dependency for codec_dahdi to only be satisfied by - DAHDI and not Zaptel, as the new transcoder interface is only in - DAHDI - -2008-08-07 18:14 +0000 [r136544] Shaun Ruffell <sruffell@digium.com> - - * codecs/codec_dahdi.c: Updated codec_dahdi to use the new - transcoder interface in the first DAHDI release. Codec dahdi no - longer functions with the transcoder interface in zaptel at this - time (which the last zaptel release was 1.4.11). NOTE: Still - needs an update to the configure script to make sure that - codec_dahdi is only built if the new transcoder interface is - present in the drivers. (Issue: DAHDI-42) - -2008-08-07 16:50 +0000 [r136488] Tilghman Lesher <tlesher@digium.com> - - * apps/app_queue.c: Update persistent state on all exit conditions. - (closes issue #12916) Reported by: sgenyuk Patches: - app_queue.patch.txt uploaded by neutrino88 (license 297) Tested - by: sgenyuk, aragon - -2008-08-07 16:30 +0000 [r136404-136484] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/doxyref.h: add a raw list of all libraries that - any part of Asterisk links directly to - - * apps/app_voicemail.c: work around a bug in gcc-4.2.3 that - incorrectly ignores the casting away of 'const' for pointers when - the developer knows it is safe to do so - - * Makefile: remove config.cache during distclean, in case the user - is using autoconf caching - -2008-08-07 01:31 +0000 [r136304-136348] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: Also, parse - useincomingcalleridonzaptransfer (and add appropriate deprecation - warnings). - - * channels/chan_dahdi.c: For backwards compatibility with previous - 1.4 versions which used "zapchan" in users.conf, ensure that we - still support it. - -2008-08-06 21:18 +0000 [r136241] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - configs/misdn.conf.sample: * The allowed_bearers setting in - misdn.conf misspelled one of its options: digital_restricted. * - Fixed some other spelling errors and typos. - -2008-08-06 20:42 +0000 [r136238] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: We only need to unregister the QueueStatus - manager command once on an unload - -2008-08-06 20:14 +0000 [r136190] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.redhat.asterisk: -C option takes a filename, - not a directory path. (closes issue #13007) Reported by: - klaus3000 - -2008-08-06 18:58 +0000 [r136168] Russell Bryant <russell@digium.com> - - * Makefile: Remove the use of --no-print-directory when compiling - subdirectories. This allows vim :make functionality to work - properly when errors have occurred in the build. Without printing - the directories, vim did not know how to find the file that the - error occurred in. If the extra bit of build noise annoys anyone, - just let me know, and I'll make this optional. - -2008-08-06 15:58 +0000 [r136062] Mark Michelson <mmichelson@digium.com> - - * main/rtp.c, channels/chan_skinny.c: Since adding the - AST_CONTROL_SRCUPDATE frame type, there are places where - ast_rtp_new_source may be called where the tech_pvt of a channel - may not yet have an rtp structure allocated. This caused a crash - in chan_skinny, which was fixed earlier, but now the same crash - has been reported against chan_h323 as well. It seems that the - best solution is to modify ast_rtp_new_source to not attempt to - set the marker bit if the rtp structure passed in is NULL. This - change to ast_rtp_new_source also allows the removal of what is - now a redundant pointer check from chan_skinny. (closes issue - #13247) Reported by: pj - -2008-08-06 03:53 +0000 [r135899-135949] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c: Fix a longstanding bug in channel walking logic, - and fix the explanation to make sense. (Closes issue #13124) - - * main/translate.c: Since powerof() can return an error condition, - it's foolhardy not to detect and deal with that condition. - (Related to issue #13240) - - * include/asterisk/threadstorage.h, include/asterisk/utils.h: 1) - Bugfix for debugging code 2) Reduce compiler warnings for another - section of debugging code (Closes issue #13237) - -2008-08-06 00:29 +0000 [r135841-135850] Mark Michelson <mmichelson@digium.com> - - * /: Remove properties that should not be here - - * apps/app_skel.c: Revert inadvertent changes to app_skel that - occurred when I was testing for a memory leak - - * include/asterisk/abstract_jb.h, main/channel.c, /, - apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h: - Merging the issue11259 branch. The purpose of this branch was to - take into account "burps" which could cause jitterbuffers to - misbehave. One such example is if the L option to Dial() were - used to inject audio into a bridged conversation at regular - intervals. Since the audio here was not passed through the - jitterbuffer, it would cause a gap in the jitterbuffer's - timestamps which would cause a frames to be dropped for a brief - period. Now ast_generic_bridge will empty and reset the - jitterbuffer each time it is called. This causes injected audio - to be handled properly. ast_generic_bridge also will empty and - reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE - frame since the change in audio source could negatively affect - the jitterbuffer. All of this was made possible by adding a new - public API call to the abstract_jb called ast_jb_empty_and_reset. - (closes issue #11259) Reported by: plack Tested by: putnopvut - -2008-08-05 23:13 +0000 [r135799] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c, - include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit - Tested by: murf I discovered that also, in the previous bug fixes - and changes, the cdr.conf 'unanswered' option is not being - obeyed, so I fixed this. And, yes, there are two 'answer' times - involved in this scenario, and I would agree with you, that the - first answer time is the time that should appear in the CDR. (the - second 'answer' time is the time that the bridge was begun). I - made the necessary adjustments, recording the first answer time - into the peer cdr, and then using that to override the bridge - cdr's value. To get the 'unanswered' CDRs to appear, I purposely - output them, using the dial cmd to mark them as DIALED (with a - new flag), and outputting them if they bear that flag, and you - are in the right mode. I also corrected one small mention of the - Zap device to equally consider the dahdi device. I heavily tested - 10-sec-wait macros in dial, and without the macro call; I tested - hangups while the macro was running vs. letting the macro - complete and the bridge form. Looks OK. Removed all the - instrumentation and debug. - -2008-08-05 21:34 +0000 [r135747] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the - meaning of the flag IAX_HASCALLERID was perverted. This change - reverts IAX2 to the original meaning, which was, that the - callerid set on the client should be overridden on the server, - even if that means the resulting callerid is blank. In other - words, if you set "callerid=" in the IAX config, then the - callerid should be overridden to blank, even if set on the - client. Note that there's a distinction, even on realtime, - between the field not existing (NULL in databases) and the field - existing, but set to blank (override callerid to blank). - -2008-08-05 13:25 +0000 [r135597] Sean Bright <sean.bright@gmail.com> - - * main/cli.c: Use PATH_MAX for filenames - -2008-08-04 20:15 +0000 [r135536] Russell Bryant <russell@digium.com> - - * configs/chan_dahdi.conf.sample: fix a config sample typo - -2008-08-04 17:07 +0000 [r135479-135482] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script - - * apps/app_voicemail.c: Memory leak on unload (closes issue #13231) - Reported by: eliel Patches: app_voicemail.leak.patch uploaded by - eliel (license 64) - -2008-08-04 16:26 +0000 [r135473] Russell Bryant <russell@digium.com> - - * configs/chan_dahdi.conf.sample: Add a minor clarification to the - documentation of mohinterpret and mohsuggest - -2008-08-01 11:43 +0000 [r135055-135058] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_ices.c: make app_ices compile on OpenBSD. - - * channels/chan_skinny.c: fix some potential deadlocks in - chan_skinny (closes issue #13215) Reported by: qwell Patches: - 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) - Tested by: mvanbaak - -2008-07-31 22:18 +0000 [r134983] Kevin P. Fleming <kpfleming@digium.com> - - * main/http.c: accomodate users who seem to lack a sense of humor - :-) - -2008-07-31 21:53 +0000 [r134976] Tilghman Lesher <tlesher@digium.com> - - * sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in - callfiles and manager, to allow video calls to be set up from - callfiles and AMI. (closes issue #9531) Reported by: Geisj - Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76 - (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by - Corydon76 (license 14) Tested by: Corydon76 - -2008-07-31 19:37 +0000 [r134915] Russell Bryant <russell@digium.com> - - * apps/app_ices.c: Get app_ices working again (closes issue #12981) - Reported by: dlogan Patches: - 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant - (license 36) 20080709__app_ices_v2_update_14.diff uploaded by - bbryant (license 36) Tested by: bbryant - -2008-07-31 19:23 +0000 [r134883] Steve Murphy <murf@digium.com> - - * res/res_features.c: (closes issue #11849) Reported by: greyvoip - Tested by: murf OK, a few days of debugging, a bunch of - instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and - 5 solid notebook pages of notes later, I have made the small - tweek necc. to get the start time right on the second CDR when: A - Calls B B answ. A hits Xfer button on sip phone, A dials C and - hits the OK button, A hangs up C answers ringing phone B and C - converse B and/or C hangs up But does not harm the scenario - where: A Calls B B answ. B hits xfer button on sip phone, B dials - C and hits the OK button, B hangs up C answers ringing phone A - and C converse A and/or C hangs up The difference in start times - on the second CDR is because of a Masquerade on the B channel - when the xfer number is sent. It ends up replacing the CDR on the - B channel with a duplicate, which ends up getting tossed out. We - keep a pointer to the first CDR, and update *that* after the - bridge closes. But, only if the CDR has changed. I hope this - change is specific enough not to muck up any current CDR-based - apps. In my defence, I assert that the previous information was - wrong, and this change fixes it, and possibly other similar - scenarios. I wonder if I should be doing the same thing for the - channel, as I did for the peer, but I can't think of a scenario - this might affect. I leave it, then, as an exersize for the - users, to find the scenario where the chan's CDR changes and - loses the proper start time. - -2008-07-31 16:45 +0000 [r134814] Russell Bryant <russell@digium.com> - - * channels/iax2-parser.c: In case we have some processing threads - that free more frames than they allocate, do not let the frame - cache grow forever. (closes issue #13160) Reported by: tavius - Tested by: tavius, russell - -2008-07-31 15:56 +0000 [r134758] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Add more timeout checks into app_queue, - specifically targeting areas where an unknown and potentially - long time has just elapsed. Also added a check to try_calling() - to return early if the timeout has elapsed instead of potentially - setting a negative timeout for the call (thus making it have *no* - timeout at all). (closes issue #13186) Reported by: - miquel_cabrespina Patches: 13186.diff uploaded by putnopvut - (license 60) Tested by: miquel_cabrespina - -2008-07-30 22:39 +0000 [r134704] Tilghman Lesher <tlesher@digium.com> - - * main/sched.c, include/asterisk/sched.h: Oops, wrong define - -2008-07-30 22:02 +0000 [r134652] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes - issue #13051) Reported by: pj This patch substitutes commas in - the expr supplied to the if () statement, as in if ( expr ) ... - This solves both the bugs above, and makes the source symmetric - with switch statements, which were earlier reported to need this - sort of treatment. I tested this using the examples, both for the - compiler and at run time. Looks good. - -2008-07-30 21:38 +0000 [r134649] Tilghman Lesher <tlesher@digium.com> - - * configure, configure.ac: Qwell pointed out, via IRC, that the - previous fix only worked when explicitly set. When nothing is - set, and the option is implied, it breaks, because configure sets - the prefix to 'NONE'. Fixing. - -2008-07-30 20:37 +0000 [r134540-134595] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max - stack size to fix a crash when compiled with LOW_MEMORY. (closes - issue #13154) Reported by: edantie - - * funcs/func_curl.c: Fix a memory leak in func_curl. Every thread - that used this function leaked an allocation the size of a - pointer. (reported by jmls in #asterisk-dev) - -2008-07-30 19:47 +0000 [r134480-134536] Tilghman Lesher <tlesher@digium.com> - - * configure, configure.ac: Only override sysconfdir and mandir when - prefix=/usr (closes issue #13093) Reported by: pabelanger - - * res/res_agi.c: launch_netscript sometimes returns -1, which fails - to set AGISTATUS. Map failure to -1, so that AGISTATUS is always - set. (closes issue #13199) Reported by: smw1218 - -2008-07-30 18:31 +0000 [r134475] Mark Michelson <mmichelson@digium.com> - - * main/app.c: Fix a spot where a function could return without - bringing a channel out of autoservice. - -2008-07-30 15:29 +0000 [r134254-134352] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: use the proper method for building version.h - - * include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c, - channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c, - apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c: - build against the now-typedef-free dahdi/user.h - -2008-07-29 15:54 +0000 [r134223] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Merging the imap_consistency branch. The - main aim of this branch was to make the IMAP code function in the - same manner as the ODBC code does, eliminating the need for so - many IMAP-specific code chunks. The focal point of all of this - work was to make the various macros (e.g. RETRIEVE, DISPOSE) - functionally equivalent. While doing the above work, I also fixed - a few bugs that I came across in my testing. Among these were 1. - Fixed message forwarding. This was completely broken when using - IMAP. 2. Fixed the inability to save new messages as old and vice - versa. 3. Fixed the "delete" options in voicemail.conf when using - IMAP storage. Even though a few bugs were fixed and the code is a - lot more consistent, the one thing that was *not* improved in - this branch was performance. The merge of this to trunk may not - come immediately due to the amount of work it will probably - involve. (closes issue #12764) Reported by: balsamcn - -2008-07-28 21:50 +0000 [r134161] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Detect when sox fails to raise the volume, - because sox can't read the file. (closes issue #12939) Reported - by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by - Corydon76 (license 14) Tested by: rickbradley - -2008-07-26 15:31 +0000 [r133980] Russell Bryant <russell@digium.com> - - * main/asterisk.c, include/asterisk/doxyref.h: Add the licensing - section to the docs in 1.4, as well, so that we can work on - having an accurate list for each version of Asterisk that is - supported - -2008-07-25 18:00 +0000 [r133649-133709] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue - #13161) - - * main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix - some errant device states by making the devicestate API more - strict in terms of the device argument (only without the unique - identifier appended). (closes issue #12771) Reported by: davidw - Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 - (license 14) Tested by: davidw, jvandal, murf - -2008-07-25 15:00 +0000 [r133578] Russell Bryant <russell@digium.com> - - * /, LICENSE: Merged revisions 133577 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) - | 2 lines Fix the IAX2 URI for calling Digium ........ - -2008-07-25 14:40 +0000 [r133572] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: We need to make sure to null-terminate the - "name" portion of SIP URI parameters so that there are no bogus - comparisons. Thanks to bbryant for pointing this out. - -2008-07-24 21:17 +0000 [r133361-133488] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes - issue #12707) - - * /: Blocked revisions 133360 via svnmerge ........ r133360 | - tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines - This part was not correctly patched for AST-2008-010. ........ - -2008-07-23 21:49 +0000 [r133295] Jason Parker <jparker@digium.com> - - * channels/chan_dahdi.c: inbandrelease is gone - it's now - inbanddisconnect - -2008-07-23 21:05 +0000 [r133226-133237] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/stringfields.h, main/utils.c: revert an - optimization that broke ABI... thanks russell! - - * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, - apps/app_dahdibarge.c, channels/chan_dahdi.c, - apps/app_dahdiras.c: make some more changes to the dahdi/zap - channel name support stuff to ensure allthe globals are 'const', - and clean up mmichelson's changes to app_chanspy to simplify the - code - -2008-07-23 19:39 +0000 [r132974-133169] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, - channels/chan_dahdi.c: As suggested by seanbright, the - PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at - compile time, since dahdi_chan_name is determined at load time. - Also changed the next_unique_id_to_use to have the static - qualifier. Also added the dahdi_chan_name_len variable so that - strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for - the suggestion. - - * apps/app_chanspy.c: Zap/pseudo is ten characters, but - DAHDI/pseudo is twelve. The strncmp call in next_channel should - account for this. - - * apps/app_chanspy.c: Update the "last" channel in next_channel in - app_chanspy so that the same pseudo channel isn't constantly - returned. related to issue #13124 - - * channels/chan_dahdi.c: Small cleanup. Move the declaration of the - DAHDI_SPANINFO variable to the block where it is used. This - allows one less #ifdef HAVE_PRI to clutter things up. Thanks to - Tzafrir for pointing this out on #asterisk-dev - - * channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI - is not defined. - -2008-07-23 15:52 +0000 [r132872-132942] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: ensure that after a channel is created, if - it happened to be in 'channel alarm' state, when that alarm - clears we won't generate a spurious 'alarm cleared' message - (closes issue #12160) Reported by: tzafrir - - * include/asterisk/stringfields.h, main/utils.c: minor optimization - for stringfields: when a field is being set to a larger value - than it currently contains and it happens to be the most recent - field allocated from the currentl pool, it is possible to 'grow' - it without having to waste the space it is currently using (or - potentially even allocate a new pool) - -2008-07-23 11:37 +0000 [r132826] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: another Fix because of r119585, this - commit has broken high frequented BRI Ports, there was a - possibility that a channel, that was marked as in_use would be - reused later, the corresponding port could got stuck then. So it - is recommended to upgrade for chan_misdn users. - -2008-07-22 22:14 +0000 [r132790] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Allow Spiraled INVITEs to work correctly - within Asterisk. Prior to this change, a spiraled INVITE would - cause a 482 Loop Detected to be sent to the caller. With this - change, if a potential loop is detected, the Request-URI is - inspected to see if it has changed from what was originally - received. If pedantic mode is on, then this inspection is fully - RFC 3261 compliant. If pedantic mode is not on, then a string - comparison is used to test the equality of the two R-URIs. This - has been tested by using OpenSER to rewrite the R-URI and send - the INVITE back to Asterisk. (closes issue #7403) Reported by: - stephen_dredge - -2008-07-22 22:11 +0000 [r132784-132787] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/options.h, main/asterisk.c, - apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c, - apps/app_dahdiras.c: fix up namespace pollution for - dahdi_chan_mode enum correct registration of AMI actions in - chan_dahdi; in zap-only mode, only register the Zap flavors of - the actions (and use Zap prefixes for headers and acks), but in - dahdi+zap mode, register both Zap and DAHDI flavors of actions - - * Makefile.rules: add rules to create preprocessor output... useful - for debugging macros - -2008-07-22 21:19 +0000 [r132713] Tilghman Lesher <tlesher@digium.com> - - * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged - revisions 132711 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) - | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ - -2008-07-22 21:17 +0000 [r132704-132712] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: ensure that if any alarms exist at channel - creation time, they are handled identically to if they occurred - later, so that later alarm clearing will work properly and 'make - sense' (closes issue #12160) Reported by: tzafrir - - * configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK - able to print a 'pretty' description of what it is doing - -2008-07-22 20:10 +0000 [r132645] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, doc/sip-retransmit.txt (added): The most - common question on the #asterisk iRC channel and on mailing lists - seems to be in regards to an error message when retransmit fails. - This is frequently misunderstood as a failure of Asterisk, not a - failure of the network to reach the other party. This document - tries to assist the Asterisk user in sorting out these issues by - explaining the logic and pointing at some possible causes. - Hopefully, we will get other questions now :-) - -2008-07-22 19:57 +0000 [r132571-132642] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: correct wording in comment - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, - include/asterisk/autoconfig.h.in, configure.ac: use renamed - libpri API call for controlling this feature (was improperly - named before) - - * channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel - on BRI spans, and don't attempt to use channel 24 as a D-channel - on spans of unexpected sizes - -2008-07-21 20:51 +0000 [r132506-132507] Brett Bryant <bbryant@digium.com> - - * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set - properly when it isn't supported on a channel (yet _another_ - useful patch by eliel). (issue #13081) Reported by: eliel - Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested - by: eliel - - * channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels - not being removed when a call was rejected (from the calling box, - not the box that denied the registration). Related to revisions - 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently - tested 1.4 with a backport from those revisions, so I didn't see - this problem (oops). - -2008-07-19 16:45 +0000 [r132311] Kevin P. Fleming <kpfleming@digium.com> - - * LICENSE: grant a license exception to allow distribution of - Asterisk binaries that use the UW IMAP Toolkit (which is licensed - under a non-GPL-compatible license) - -2008-07-18 19:06 +0000 [r131970-132112] Tilghman Lesher <tlesher@digium.com> - - * main/say.c: Fix for Taiwanese number syntax (closes issue #12319) - Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch - uploaded by CharlesWang (license 444) - - * main/config.c: Textual clarification (closes issue #13106) - Reported by: flefoll Patches: - config.c.br14.120173.patch-unknown-directive uploaded by flefoll - (license 244) - - * include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within - the destroy, to allow a scheduled job to continue, if it's - waiting on the mutex which the destroy thread has. - - * main/sched.c: Oops - - * main/sched.c, include/asterisk/sched.h: Preserve ABI - compatibility with last change - - * main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c: - Make the ast_assert call within ast_sched_del report something - useful. - -2008-07-18 16:15 +0000 [r131921] Kevin P. Fleming <kpfleming@digium.com> - - * main/dlfcn.c (removed), main/loader.c, main/Makefile, - include/asterisk/dlfcn-compat.h (removed): remove the dlfcn - compatibility stuff, because no platforms that Asterisk currently - runs on it use it, and it doesn't build anyway - -2008-07-18 15:34 +0000 [r131915] Brett Bryant <bbryant@digium.com> - - * res/res_features.c: Fix a bug in blind transfers where the - BLINDTRANSFER variable isn't always set to the other end of the - blind transfer. (closes issue #12586) - -2008-07-17 20:35 +0000 [r131790] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965) - as it appears that it was in error. This should fix talk call - progress on analog lines. (closes issue #12178) Reported by: - michael-fig Patches: 20080717__bug12178.diff.txt uploaded by - Corydon76 (license 14) - -2008-07-16 22:17 +0000 [r131491] Brett Bryant <bbryant@digium.com> - - * channels/chan_iax2.c: Fix a bug in iax2 registration that allowed - peers to register with case-insensitive names (user_cmp_cb and - peer_cmp_cb are now both case-sensitive). (closes issue #13091) - -2008-07-16 21:46 +0000 [r131480] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Apparently, in certain cases, a callno is - already destroyed when iax2_destroy is called. - -2008-07-16 20:47 +0000 [r131421] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Always ensure that the channel's tech_pvt - reference is NULL after calling the destroy callback. (closes - issue #13060) Reported by: jpgrayson Patches: - chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license - 492) - -2008-07-16 20:23 +0000 [r131299-131369] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Move the init_queue call back to where it used - to be (changed Sept 12 last year). It was moved then to prevent a - memory leak. Since then, the same memory leak recurred and was - fixed in a better way. Now it has been found that the placement - of this init_queue call can cause problems if a realtime queue - has values changed to an empty string. The problem is that the - default value for that queue parameter would not be set. (closes - issue #13084) Reported by: elbriga - - * apps/app_queue.c: Apparently, "thread safety" is important, - whatever that means. :P (Thanks Russell!) - - * apps/app_queue.c: Make absolutely certain that the transfer - datastore is removed from the calling channel once the caller is - finished in the queue. This could have weird con- sequences when - dialing local queue members when multiple transfers occur on a - single call. Also fixed a memory leak that would occur when an - attended transfer occurred from a queue member. (closes issue - #13047) Reported by: festr - -2008-07-16 17:53 +0000 [r131242] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The - problem was that, esoteric as it is, because the hangerupper - context immediately preceded the std-priv-extent macro, that the - checking code accidentally would fall from traversing hangerupper - into the std-priv-exten macro, where it would hit the hangerupper - in the 'includes', and proceed into an infinite recursion. A - small fix to traverse into the statements of the context instead - of the context solves this issue. I also added some commented out - printfs for debug, which were pretty handy in the face of a dorky - gdb. This was a problem around since the package was first - written; but evidently pretty rare in turning up in the field. - -2008-07-15 17:47 +0000 [r131012] Michiel van Baak <michiel@vanbaak.info> - - * main/cdr.c: remove 4 lines of redundant code. (closes issue - #13080) Reported by: gknispel_proformatique Patches: - trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) - -2008-07-15 17:19 +0000 [r130889-130959] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, channels/chan_sip.c: astman_send_error does not - need a newline appended -- the API takes care of that for us. - (closes issue #13068) Reported by: gknispel_proformatique - Patches: asterisk_1_4_astman_send.patch uploaded by gknispel - (license 261) asterisk_trunk_astman_send.patch uploaded by - gknispel (license 261) - - * channels/chan_iax2.c: Override the callerid in all cases when the - callerid is set in the user, not just when a remote callerid is - set. Also, if not set in the user, allow the remote CallerID to - pass through. (closes issue #12875) Reported by: dimas Patches: - 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) - -2008-07-14 17:50 +0000 [r130792] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in - app_dial to be sure there are no audiohooks present on the - channels involved. This fixed a one-way audio situation I had in - my test setup. I couldn't find any open issues that suggested - one-way audio with regards to mixmonitor (or other audiohook) - usage, though. - -2008-07-14 17:10 +0000 [r130735] Michiel van Baak <michiel@vanbaak.info> - - * main/dnsmgr.c: notify the user that dnsmgr refresh wont work when - dnsmgr is not enabled. Previously this command would - automagically appear and disappear. This was confusing. (closes - issue #12796) Reported by: chappell Patches: - dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: - russell, chappell, mvanbaak - -2008-07-14 10:38 +0000 [r130634] Russell Bryant <russell@digium.com> - - * main/audiohook.c: Bump up the debug level for a message. - -2008-07-13 22:48 +0000 [r130573] Michiel van Baak <michiel@vanbaak.info> - - * main/manager.c: fix memory leak when originate from manager - cannot create a thread (closes issue #13069) Reported by: - gknispel_proformatique Patches: - asterisk_trunk_action_originate.patch uploaded by gknispel - (license 261) Tested by: gknispel_proformatique, mvanbaak - -2008-07-13 17:56 +0000 [r130514] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Reverting 2 changesets, as it breaks - incoming IAX2 calls (Related to issue #12963) Reported by: - mvanbaak - -2008-07-12 10:25 +0000 [r130373] Michiel van Baak <michiel@vanbaak.info> - - * pbx/pbx_ael.c: in 1.4 the functions still have | as argument - seperator. This commit fixes the use of RAND in the ael random - function. (closes issue #13061) Reported by: danpwi - -2008-07-11 22:23 +0000 [r130298-130317] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: forcibly remove the modules that are changing names - - * include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c, - Makefile, main/channel.c, apps/app_dahdibarge.c, - channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c, - apps/app_dahdiras.c, main/file.c, - contrib/utils/zones2indications.c, include/asterisk/channel.h, - channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility - work, with lots more to come... this tree is not yet ready for - users to be easily upgrading or switching, but it needs to be :-) - -2008-07-11 20:03 +0000 [r130173-130236] Mark Michelson <mmichelson@digium.com> - - * main/audiohook.c: Remove redundant logic - - * main/audiohook.c: Fix a typo in audiohook_read_frame_both. While - this change has not been proven to fix any specific issue, it is - incorrect and could cause unforeseen problems. - -2008-07-11 18:51 +0000 [r130102-130169] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Ensure that a destination callno of 0 will - not match for frames that do not start a dialog (new, lagrq, and - ping). (closes issue #12963) Reported by: russellb Patches: - chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) - - * channels/chan_agent.c: Pass the devicestate from an underlying - channel up through the Agent channel. This should make the Agent - always report the correct device state, even when the underlying - channel is used for other purposes. (closes issue #12773) - Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded - by Corydon76 (license 14) Tested by: davidw - -2008-07-11 16:08 +0000 [r130039-130042] Kevin P. Fleming <kpfleming@digium.com> - - * doc/configuration.txt, configs/extensions.conf.sample, - configs/sla.conf.sample, configs/zapata.conf.sample (removed), - contrib/scripts/autosupport, README, - configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c, - include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt, - configs/extensions.ael.sample, configs/smdi.conf.sample: new - installations should be using DAHDI instead of Zaptel, so the - sample config file is now chan_dahdi.conf instead of zapata.conf - also, convert remaining references to zapata.conf in various - places - - * configs/zapata.conf.sample, channels/chan_dahdi.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: add support for a - configuration parameter for 'inband audio during RELEASE', which - is currently mandatory in libpri-1.4.4 but will become - configurable in libpri-1.4.5 later today (related to issue - #13042) - -2008-07-11 14:18 +0000 [r129970] Russell Bryant <russell@digium.com> - - * include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro - ... - -2008-07-11 14:14 +0000 [r129907-129967] Kevin P. Fleming <kpfleming@digium.com> - - * main/astmm.c: simplify calculation - - * main/astmm.c: fix a flaw found while experimenting with structure - alignment and padding; low-fence checking would not work properly - on 64-bit platforms, because the compiler was putting 4 bytes of - padding between the fence field and the allocation memory block - added a very obvious runtime warning if this condition reoccurs, - so the developer who broke it can be chastised into fixing it :-) - - * sounds/Makefile: don't attempt to set user/group ownership of - extracted sound files (reported on asterisk-users) (closes issue - #13059) - -2008-07-10 21:57 +0000 [r129741-129803] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Correctly deal with duplicate NEW frames - (due to retransmission). Also, fixup the destination call number - matching to be more strict and reliable. (closes issue #12963) - Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch - uploaded by jpgrayson (license 492) Tested by: jpgrayson, - Corydon76 - - * res/res_config_odbc.c: Oops - -2008-07-10 16:03 +0000 [r129567] Russell Bryant <russell@digium.com> - - * sample.call: Note that pbx_spool.so is the module used for call - files (inspired by a question in #asterisk) - -2008-07-10 13:57 +0000 [r129505] Sean Bright <sean.bright@gmail.com> - - * main/editline: Update svn:ignore - -2008-07-09 19:32 +0000 [r129436] Mark Michelson <mmichelson@digium.com> - - * main/rtp.c: Fix a problem where inbound rfc2833 audio would be - sent to the core instead of being P2P bridged. When the core - regenerated the rfc2833 packet for the outbound leg, the SSRC - would be different than the RTP audio on the call leg causing - DTMF detection issues on the far end. (closes issue #12955) - Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by - tsearle (license 373) Tested by: tonyredstone - -2008-07-09 13:41 +0000 [r129343] Sean Bright <sean.bright@gmail.com> - - * main/editline/makelist (removed), main/editline/makelist.in - (added), main/editline/configure, main/editline/Makefile.in, - main/editline/configure.in: Look for the system installed awk - instead of assuming it's at /usr/bin/awk. Pointed out by jmls via - #asterisk-dev. - -2008-07-08 21:31 +0000 [r129158-129208] Mark Michelson <mmichelson@digium.com> - - * doc/imapstorage.txt: Update documentation to have the correct - option name - - * apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related - timeouts to IMAP voicemail in 1.4 since it should solve bugs - people are experiencing. Specifically, there are times where - communication with the IMAP server causes system calls to block - forever. If this should happen when querying the mailbox so that - chan_sip's do_monitor thread can send MWI to a phone, it means - that SIP calls cannot be processed any more. The timeout options - are outlined in doc/imapstorage.txt. Defaults for the timeouts - are sixty seconds. (closes issue #12987) Reported by: mthomasslo - -2008-07-08 20:27 +0000 [r129047-129149] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h: - Cause SIP to return a 480 instead of a 404 when a sip peer - exists, but is not registered. (closes issue #12885) Reported by: - ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 - (license 14) Tested by: ibc - - * channels/chan_iax2.c: Timestamp decoding for video mini-frames is - bogus, because the timestamp only includes 15 bits, unlike voice - frames, which contain a 16-bit timestamp. (closes issue #13013) - Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch - uploaded by jpgrayson (license 492) - -2008-07-08 09:52 +0000 [r128912-128950] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't hangup the call if we can't resolve - the Contact if there's a proxy route set for the call. ---- This - comment was added a while ago and today it hit me badly. /* OEJ: - Possible issue that may need a check: If we have a proxy route - between us and the device, should we care about resolving the - contact or should we just send it? */ - - * channels/chan_sip.c: Fix issues where repeated messages where - ignored, but retransmitted reliably instead of unreliably. - Reported by: johan Patches: 12746.txt uploaded by oej (license - 306) Tested by: johan (issue #12746) - -2008-07-08 00:01 +0000 [r128812-128856] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Check for non-NULL before stripping - characters. (closes issue #12954) Reported by: bfsworks Patches: - 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) - Tested by: deti - - * apps/app_voicemail.c: Stop using deprecated method, as requested - by Kevin. - -2008-07-07 22:41 +0000 [r128795] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix handling of when a pvt disappears. - Properly return the pvt locked and don't hold the pvt lock while - destroying the ast_channel. (closes issue #13014) Reported by: - jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by - jpgrayson (license 492) - -2008-07-07 20:47 +0000 [r128737] Sean Bright <sean.bright@gmail.com> - - * channels/chan_iax2.c: Remove spurious trailing whitespace from - log messages and fix a spelling error in a log message. (closes - issue #13017) Reported by: jpgrayson Patches: - chan_iax2_space_after_newline.patch uploaded by jpgrayson - (license 492) chan_iax2_spelling.patch uploaded by jpgrayson - (license 492) - -2008-07-07 17:02 +0000 [r128639] Mark Michelson <mmichelson@digium.com> - - * channels/chan_iax2.c: By using the iaxdynamicthreadcount to - identify a thread, it was possible for thread identifiers to be - duplicated. By using a globally-unique monotonically- increasing - integer, this is now avoided. (closes issue #13009) Reported by: - jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by - jpgrayson (license 492) - -2008-07-07 16:51 +0000 [r128637] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: use tzafrir's patch to fix this problem - properly... i made the previous set of changes without thoroughly - testing them, doh! (closes issue #12911) Reported by: tzafrir - Patches: custum_dahdi_configure_2.diff uploaded by tzafrir - (license 46) Tested by: tzafrir - -2008-07-04 16:11 +0000 [r127973-128029] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_config.c: Move the free down one - - * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the - 'dialplan remove extension' logic, so that it a) works with - cidmatch, and b) completes contexts correctly when the extension - is ambiguous. (closes issue #12980) Reported by: licedey Patches: - 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76 - -2008-07-03 22:20 +0000 [r127754-127895] Kevin P. Fleming <kpfleming@digium.com> - - * apps/Makefile: remove this, it has been moved to the main - Makefile - - * Makefile, main/editline/np/vis.c: a couple of small - Solaris-related fixes (closes issue #11885) Reported by: snuffy, - asgaroth - - * configure, main/Makefile, configure.ac, acinclude.m4: ensure that - DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places - needed improve AST_EXT_LIB_CHECK to accept (and remember) - additional CFLAGS data like it does in trunk already (closes - issue #12911) Reported by: tzafrir - -2008-07-03 00:16 +0000 [r127663] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, - channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h: - The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported - by: murf Tested by: murf, deeperror (closes issue #12907) - Reported by: falves11 Tested by: murf, falves11 (closes issue - #11849) Reported by: greyvoip As to 11849, I think these changes - fix the core problems brought up in that bug, but perhaps not the - more global problems created by the limitations of CDR's - themselves not being oriented around transfers. Reopen if necc, - but bug reports are not the best medium for enhancement - discussions. We need to start a second-generation CDR - standardization effort to cover transfers. (closes issue #11093) - Reported by: rossbeer Tested by: greyvoip, murf - -2008-07-02 20:47 +0000 [r127560] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Fix thread-safety of some of the - pbx_builtin_getvar_helper calls - -2008-07-02 19:47 +0000 [r127501] Tilghman Lesher <tlesher@digium.com> - - * main/acl.c: Merged revisions 127466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | - tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines - Solaris fix (closes issue #12949) Reported by: snuffy Patches: - bug_12949.diff uploaded by snuffy (license 35) ........ - -2008-07-01 23:36 +0000 [r127244] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Add error message to failed open(2) calls - inside the copy() function of app_voicemail. This idea came as - part of my work in helping to resolve issue #12764. - -2008-07-01 20:25 +0000 [r126999-127133] Tilghman Lesher <tlesher@digium.com> - - * build_tools/cflags.xml, channels/chan_iax2.c: Disable the old, - slow search for matching callno in chan_iax2 (but allow it to be - reenabled for debugging) - - * channels/chan_iax2.c: Oops - - * channels/chan_iax2.c: Change around how we schedule pings and - lagrqs, and fix a reason why the jobs were not getting properly - cancelled. (closes issue #12903) Reported by: stevedavies - Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76 - (license 14) Tested by: stevedavies - - * channels/chan_iax2.c: Suppress annoying warning by finding the - remaining cases where the callno is not in the hash. - -2008-07-01 14:59 +0000 [r126735-126902] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Use domain part of SIP uri in register= - configuration as fromdomain. Reported by: one47 Patches: - sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes - issue #12474) - - * channels/chan_sip.c: Handle escaped URI's in call pickups. Patch - by oej and IgorG. Reported by: IgorG Patches: - bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by: - IgorG, oej (closes issue #12299) - - * configs/sip.conf.sample: Clear up documentation on "domain=" - setting in sip.conf Reported by: davidw (closes issue #12413) - - * channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs - requests (to confirm that the dialog exist). Don't bother - checking the request URI. (closes issue #11264) Reported by: ibc - - * channels/chan_sip.c: Fix bad XML for hold notification. Reported - by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432) - (closes issue #12942) - -2008-06-30 23:11 +0000 [r126680] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Load the proper channel configuration file - based on which driver was detected. - -2008-06-30 22:30 +0000 [r126674] Tilghman Lesher <tlesher@digium.com> - - * configs/zapata.conf.sample: Add note about other names for - EuroISDN - -2008-06-30 16:05 +0000 [r126573] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS - version of the recently added DEADLOCK_AVOIDANCE() macro. This - caused the lock to not actually be released, and as a result, not - avoid deadlocks at all. This resolves the issues reported in the - last while about Asterisk locking up all over the place (and most - commonly, in chan_iax2). (closes issue #12927) (closes issue - #12940) (closes issue #12925) (potentially closes others ...) - -2008-06-30 12:50 +0000 [r126516] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Send all responses to an INVITE reliably, so - that we retransmit if we don't get an ACK and also fail if we - don't get the very same precious ACK. Based on patch by tsearle, - with my own additions. (closes issue #12951) Reported by: tsearle - Patches: busy_retransmit.patch uploaded by tsearle (license 373) - -2008-06-29 18:05 +0000 [r126395] Kevin P. Fleming <kpfleming@digium.com> - - * pbx/Makefile: ignore warnings for prototypes in GTK headers - -2008-06-27 22:01 +0000 [r125740-126056] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: When we get a 408 Timeout, don't stop trying - to re-register. (closes issue #12863) Reported by: ricvil - - * include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined - to HAVE_ZAPTEL in dahdi_compat.h, we must first check for - HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches: - tonezone_compat.diff uploaded by opticron (license 267) - - * main/utils.c, include/asterisk/lock.h: In this debugging - function, copy to a buffer instead of using potentially unsafe - pointers. - - * channels/chan_local.c: Add proper deadlock avoidance. (closes - issue #12914) Reported by: ozan Patches: - 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14) - Tested by: ozan - -2008-06-26 23:03 +0000 [r125587] Jason Parker <jparker@digium.com> - - * main/utils.c: Make sure to unlock the lock_info lock (huh?). - Possible deadlock? - -2008-06-26 22:52 +0000 [r125476-125585] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Add the interface of a queue member to the - output of the "queue show" command so that it can easily be - associated with a queue member's name. This helps so that the - appropriate queue member can be removed or paused since the - interface is required, not the member's name. (closes issue - #12783) Reported by: davevg Patches: app_queue.diff uploaded by - davevg (license 209) with small mod from me - - * apps/app_queue.c: Backport of attended transfer queue_log patch - from trunk. This patch allows for attended transfers to be logged - in the queue_log the same way that blind transfers have always - been. It was decided by popular opinion on the asterisk-dev - mailing list that this should be backported to 1.4. Thanks to - everyone who gave an opinion. - - * apps/app_queue.c: Prior to this patch, the "queue show" command - used cached information for realtime queues instead of giving - up-to-date info. Now realtime is queried for the latest and - greatest in queue info. (closes issue #12858) Reported by: bcnit - Patches: queue_show.patch uploaded by putnopvut (license 60) - -2008-06-26 16:32 +0000 [r125384] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add support for peer realm based auth (a few - missing lines, the rest is well documented but never worked) - -2008-06-26 15:30 +0000 [r125327] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: ensure that (whenever possible) if we - generate a log message because an ioctl() call to DAHDI/Zaptel - failed, that we include the reason it failed by including the - stringified error number (issue AST-80) - -2008-06-26 11:01 +0000 [r125218-125276] Tilghman Lesher <tlesher@digium.com> - - * main/rtp.c: Check for rtcp structure before trying to delete - schedule. (closes issue #12872) Reported by: destiny6628 Patches: - 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) - Tested by: destiny6628 - - * configs/agents.conf.sample: Document ackcall=always. (closes - issue #12852) Reported by: davidw - -2008-06-25 22:21 +0000 [r125132] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_rpt.c, include/asterisk/dahdi_compat.h, - channels/chan_dahdi.c, configure, - include/asterisk/tonezone_compat.h (added), configure.ac: allow - tonezone to live in a different place than DAHDI/Zaptel, since - dahdi-tools and dahdi-linux are now separate packages and can be - installed in different places don't include tonezone.h in - dahdi_compat.h, because only a couple of modules need it get - app_rpt building again after the DAHDI changes (closes issue - #12911) Reported by: tzafrir - -2008-06-25 00:46 +0000 [r124908-124965] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: Pvt deadlock causes some channels to get - stuck in Reserved status. (closes issue #12621) Reported by: - fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by - Corydon76 (license 14) Tested by: fabianoheringer - - * apps/app_voicemail.c: Occasionally control characters find their - way into CallerID. These need to be stripped prior to placing - CallerID in the headers of an email. (closes issue #12759) - Reported by: RobH Patches: 20080602__bug12759__2.diff.txt - uploaded by Corydon76 (license 14) Tested by: RobH - - * channels/chan_sip.c: Don't access the pvt structure if unable to - acquire the lock. (closes issue #12162) Reported by: norman - Patches: 12162-lockfail.diff uploaded by qwell (license 4) - -2008-06-23 21:22 +0000 [r124743] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: emit a warning if the old IAX2 call - searching code finds a call when the new code did not... so that - we can get rid of the old code in 2-3 months - -2008-06-22 02:54 +0000 [r124540] Steve Murphy <murf@digium.com> - - * apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac - Sorry, my testing did not contain the simple case of forkCDR(v), - I am much embarrassed to admit. If I had, I would have more - solidly initialized the opts element for varset. - -2008-06-20 23:12 +0000 [r124395-124450] Tilghman Lesher <tlesher@digium.com> - - * apps/app_rpt.c: usleep with a value over 1,000,000 is - nonportable. Changing to use sleep() instead. (closes issue - #12814) Reported by: pputman Patches: app_rtp_sleep.patch - uploaded by pputman (license 81) - - * main/app.c: If the last character in a string to be parsed is the - delimiter, then we should count that final empty string as an - additional argument. - -2008-06-20 21:14 +0000 [r124372] Jeff Gehlbach <jeffg@opennms.org> - - * doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in - digium-mib.txt and asterisk-mib.txt to placate smilint - bug - 12905 - -2008-06-20 20:16 +0000 [r124182-124315] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c: When using a Local channel, started by a - call file, with a destination of an AGI script, the AGI script - does not always get notified of a hangup if the underlying - channel hangs up early. (closes issue #11833) Reported by: IgorG - Patches: local_hangup-v1.diff uploaded by IgorG (license 20) - - * channels/chan_dahdi.c: It's possible for a hangup to be received, - even just after the initial cid spill. (closes issue #12453) - Reported by: Alex728 Patches: 20080604__bug12453.diff.txt - uploaded by Corydon76 (license 14) - -2008-06-19 20:28 +0000 [r124112] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix IMAP forwarding so that messages are - sent to the proper mailbox. (closes issue #12897) Reported by: - jaroth Patches: destination_forward.patch uploaded by jaroth - (license 50) - -2008-06-19 19:55 +0000 [r124066] Brett Bryant <bbryant@digium.com> - - * main/utils.c: Merge revision 124064 from trunk. Add errors that - report any locks held by threads when they are being closed. - -2008-06-19 16:58 +0000 [r123710-123930] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c: Change informative messages to use the _multiple - variant when multiple formats are possible. (Closes issue #12848) - Reported by klaus3000 - - * build_tools/strip_nonapi: Only process 40 arguments (20 files) at - once with xargs, because some older shells may force xargs to - separate on an odd boundary. (Closes issue #12883) Reported by - Nik Soggia - - * configs/sip.conf.sample: Correct description of notifyringing - option. (Closes issue #12890) Reported by gminet - - * main/asterisk.c: The RDTSC instruction was introduced on the - Pentium line of microprocessors, and is not compatible with - certain 586 clones, like Cyrix. Hence, asking for i386 - compatibility was always incorrect. See - http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported - by tecnoxarxa - - * main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add - support for saying numbers in Hebrew. (closes issue #11662) - Reported by: greenfieldtech Patches: say.c.patch-12042008 - uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods - uploaded by greenfieldtech (with signficant changes to the - spreadsheet by me) - - * pbx/pbx_spool.c: Set the variables top-down, so that if a script - sets a variable more than once, the last one will take - precedence. (closes issue #12673) Reported by: phber Patches: - 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) - -2008-06-17 20:26 +0000 [r123485] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Make chan_sip build under dev mode with - compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this - -2008-06-17 18:56 +0000 [r123391] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Fix 3 more places where not locking the - structure could cause the wrong lock to be unlocked. (Closes - issue #12795) - -2008-06-17 18:09 +0000 [r123274-123333] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Cisco BTS sends SIP responses with a tab - between the Cseq number and SIP request method in the Cseq: - header. Asterisk did not handle this properly, but with this - patch, all is well. (closes issue #12834) Reported by: tobias_e - Patches: 12834.patch uploaded by putnopvut (license 60) Tested - by: tobias_e - - * apps/app_queue.c: davidw pointed out that the holdtime - calculation used by app_queue does not use "boxcar" filtering as - the comments say. The term "boxcar" means that the number of - samples used to calculate stays constant, with new samples - replacing the oldest ones. The queue holdtime calculation uses - all holdtime samples collected since the queue was loaded, so the - comment has been changed to be accurate. (closes issue #12781) - Reported by: davidw - -2008-06-17 15:48 +0000 [r123271] Russell Bryant <russell@digium.com> - - * main/astobj2.c: Fix a memory leak in astobj2 that was pointed out - by seanbright. When a container got destroyed, the underlying - bucket list entry for each object that was in the container at - that time did not get free'd. - -2008-06-16 19:50 +0000 [r123110-123113] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_mgcp.c, channels/chan_dahdi.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_iax2.c: Port "hasvoicemail" change from SIP to - other channel drivers - - * channels/chan_sip.c: People expect that if "hasvoicemail" is set - in users.conf, even if "mailbox" isn't set, that SIP will detect - a mailbox. (closes issue #12855) Reported by: PLL Patches: - 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: PLL - -2008-06-16 12:31 +0000 [r122869-122919] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only compare the first 15 characters so that - even if the charset is specified we still accept it as SDP. - (closes issue #12803) Reported by: lanzaandrea Patches: - chan_sip.c.diff uploaded by lanzaandrea (license 496) - - * channels/chan_sip.c: Don't send a BYE on a dialog that is already - gone during a REFER. (closes issue #12865) Reported by: flefoll - Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by - flefoll (license 244) - -2008-06-13 21:44 +0000 [r122713] Mark Michelson <mmichelson@digium.com> - - * main/autoservice.c: Short circuit the loop in autoservice_run if - there are no channels to poll. If we continued, then the result - would be calling poll() with a NULL pollfd array. While this is - fine with POSIX's poll(2) system call, those who use Asterisk's - internal poll mechanism (Darwin systems) would have a failed - assertion occur when poll is called. (related to issue #10342) - -2008-06-13 18:57 +0000 [r122663] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed - dahdi compatability header from assuming either dahdi or zaptel - is installed (may not have either) - -2008-06-13 17:45 +0000 [r122617] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c: Remove extra option from previous solution - attempt - -2008-06-13 17:36 +0000 [r122613] Jeff Peeler <jpeeler@digium.com> - - * configure, configure.ac: (closes issue #12846) Reported by: - Netview Tested by: jpeeler Use correct location to search for - tonezone. - -2008-06-13 16:29 +0000 [r122589] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c, res/res_features.c: This should fix the behavior - of the 'T' dial feature being passed incorrectly to the - transferee when builtin_atxfers are used. Also, doing a - builtin_atxfer to parking was broken and is fixed here as well. - (closes issue #11898) Reported by: sergee Tested by: otherwiseguy - -2008-06-12 19:08 +0000 [r122314] Jeff Peeler <jpeeler@digium.com> - - * main/indications.c, include/asterisk/dahdi_compat.h (added), - main/loader.c, main/channel.c, channels/chan_dahdi.c (added), - configure, apps/app_zapscan.c (removed), apps/app_zapras.c - (removed), main/app.c, include/asterisk/options.h, - apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c, - channels/chan_zap.c (removed), apps/app_page.c, - include/asterisk/indications.h, apps/app_dahdiras.c (added), - configure.ac, apps/app_disa.c, include/asterisk/channel.h, - apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c - (removed), channels/chan_misdn.c, apps/app_flash.c, - build_tools/menuselect-deps.in, funcs/func_channel.c, - main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c, - codecs/codec_dahdi.c (added), res/res_indications.c, - pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c, - main/asterisk.c, apps/app_dahdibarge.c (added), - apps/app_meetme.c, include/asterisk/autoconfig.h.in, - apps/app_dahdiscan.c (added), acinclude.m4, - res/res_musiconhold.c, codecs/codec_zap.c (removed), - channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI - usage favored, but all Zap stuff should continue working. Release - announcement to follow. - -2008-06-12 18:50 +0000 [r122311] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Properly play a holdtime message if the - announce-holdtime option is set to "once." (closes issue #12842) - Reported by: ramonpeek Patches: patch001.diff uploaded by - ramonpeek (license 266) - -2008-06-12 18:22 +0000 [r122259] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix some race conditions that cause - ast_assert() to report that chan_iax2 tried to remove an entry - that wasn't in the scheduler - -2008-06-12 15:46 +0000 [r122208] Jeff Peeler <jpeeler@digium.com> - - * apps/app_parkandannounce.c, res/res_features.c: (closes issue - #12193) Reported by: davidw Patch by: Corydon76, modified by me - to work properly with ParkAndAnnounce app - -2008-06-12 15:18 +0000 [r122130-122137] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c: Flipflop the sections for two options, since - the section for 'X' (exit context) may otherwise absorb - keypresses meant for 's' (admin/user menu). (closes issue #12836) - Reported by: blitzrage Patches: 20080611__bug12836.diff.txt - uploaded by Corydon76 (license 14) Tested by: blitzrage - - * main/channel.c: Occasionally, the alertpipe loses its nonblocking - status, so detect and correct that situation before it causes a - deadlock. (Reported and tested by ctooley via #asterisk-dev) - -2008-06-12 14:51 +0000 [r122127] Steve Murphy <murf@digium.com> - - * main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the - code added to ast_cdr_busy, _failed, and _noanswer was redundant. - Didn't spot it until I was resolving conflicts in trunk. Ugh. - Redundant code removed. It wasn't harmful. Just dumb. - -2008-06-12 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.21 released. - -2008-06-06 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.21-rc2 released. - -2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell@digium.com> - - * UPGRADE-1.2.txt: fix filename - - * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file - from Asterisk 1.2, for handy reference. - -2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Ignore appended resource when comparing JIDs. - -2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell@digium.com> - - * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the - forwarding station isn't very useful for most people, so pull in - the changes that allow searching for SMDI messages based on other - components of the SMDI message. Also, update the SMDI - documentation. - -2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Make sure that the string we set will survive - the unref of the queue member. Thanks to Russell, who pointed - this out. - -2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c: If we fail to setup the PRI request channel, - don't continue, exit with an error. (closes issue #11989) - Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt - uploaded by Corydon76 (license 14) - -2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: Make the "dialplan remove include" CLI command - actually work. Also, tweak some formatting, and make the success - message a little bit more clear. (closes AST-52) - -2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Tab completion when removing a member should - give the member's interface, not the name, since the interface is - what is expected for the command. (closes issue #12783) Reported - by: davevg - -2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message - for when the dialplan is done reloading. (closes issue #12716) - Reported by: chappell Patches: dialplan_reload_2.diff uploaded by - chappell (license 8) - -2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_loopback.c: Due to incorrect use of the - AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform - any translation on the extension number before searching for it - in the target context. (closes issue #12473) Reported by: - chappell Patches: pbx_loopback.c.diff uploaded by chappell - (license 8) - -2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler@digium.com> - - * main/config.c: (closes issue #11594) Reported by: yem Tested by: - yem This change decreases the buffer size allocated on the stack - substantially in config_text_file_load when LOW_MEMORY is turned - on. This change combined with the fix from revision 117462 - (making mkintf not copy the zt_chan_conf structure) was enough to - prevent the crash. - -2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix another place where peer->callno could - change at a very bad time, and also fix a place where a peer was - used after the reference was released. (inspired by rev 120001) - -2008-06-03 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.21-rc1 released. - -2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: When listing the manager users, managers in - users.conf are not shown, even though they are allowed to - connect. (closes issue #12594) Reported by: bkruse Patches: - 12594-managerusers-2.diff uploaded by qwell (license 4) Tested - by: bkruse - - * channels/chan_iax2.c: Save the callno when we're poking, because - our peer structure could change during deadlock avoidance (and - thus we unlock the wrong callno, causing a cascade failure). - (closes issue #12717) Reported by: gewfie Patches: - 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) - Tested by: gewfie - -2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ref.ael-vtest17, - pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, - pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL. - Hadn't updated this for the changes I made to preserve ${EXTEN} - in switches, which affected several tests because it adds extra - priorities, and at least one needed to be updated because of the - removal of the empty extension warning message. - - * pbx/pbx_ael.c: as per - http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, - which is a message from Philipp Kempgen, requesting that the - WARNING that an extension is empty be reduced to a NOTICE or - less, as empty extensions are syntactically possible, and no big - deal. With which I agree, and have removed that WARNING message - entirely. I think it is not necessary to see this message. It - didn't state that a NoOp() was inserted automatically on your - behalf, and really, as users, who cares? Why freak out dialplan - writers with unnecessary warnings? The details of the - machinations a compiler goes thru to produce working assembly - code is of little interest to most programmers-- we will follow - the unix principal of doing our work silently. - -2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Treat ECONNREFUSED as an error that will - stop further retransmissions. (issue #AST-58, patch from - Switchvox) - -2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Revert a change made for issue #12479. This - change caused a regression such that a dial string such as - (IAX2/foo) did not automatically fall back to dialing the 's' - extension anymore. (closes issue #12770) Reported by: dagmoller - - * main/manager.c: Improve CLI command blacklist checking for the - command manager action. Previously, it did not handle case or - whitespace properly. This made it possible for blacklisted - commands to get executed anyway. (closes issue #12765) - -2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest - account with any configured XMPP client (in jabber.conf). The - actual connection is made when a call comes in Asterisk. Fix the - ast_aji_get_client function that was not able to retrieve an XMPP - client from its JID. (closes issue #12085) Reported by: junky - Tested by: phsultan - -2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get - sent because it comes up too soon, make sure to reschedule so it - gets sent later. - -2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is - enabled - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added - counter for unhandled_bmsg Print, this prevents the logs to be - flooded to fast and save CPU in this error scenario. Added - 'last_used' element to bc structure, when a bchannel changes from - used to free this exact time will be marked in last_used. When a - new channel is requested the find_free_chan function will check - if the new empty channel was used within the last second, if yes - it will search for the next channel, if no it will return this - channel. This simple mechanism has prooven to prevent race - conditions where the NT and TE tried to allocate the exact same - channel at the same time (RELEASE cause: 44). - -2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Change a debug message to an actual debug - message - - * apps/app_dial.c: Fix another typo in documentation - -2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_dial.c: small typo fix 'retires' => 'retries' - -2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher@digium.com> - - * apps/app_queue.c: When joinempty=strict, it only failed on join - if there were busy members. If all members were logged out OR - paused, then it (incorrectly) let callers join the queue. (closes - issue #12451) Reported by: davidw - -2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp@digium.com> - - * main/autoservice.c: Fix a bug I found while testing for another - issue. - -2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel@vanbaak.info> - - * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, - 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.slackware.asterisk: dont use a bashism way to - check the $VERSION variable. The rc/init.d scripts, and - safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2 - (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski) - (closes issue #12687) Reported by: loloski Patches: - 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by - mvanbaak (license 7) Tested by: loloski, mvanbaak - -2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 - May 2008) | 7 lines - Instead of only enforcing destination call - number checking on an ACK, check all full frames except for PING - and LAGRQ, which may be sent by older versions too quickly to - contain the destination call number. (As suggested by Tim Panton - on the asterisk-dev list) - Merge changes from - team/russell/iax2-frame-race, which prevents PING and LAGRQ from - being sent before the destination call number is known. ........ - - * main/autoservice.c: Fix a race condition in channel autoservice. - There was still a small window of opportunity for a DTMF frame, - or some other deferred frame type, to come in and get dropped. - (closes issue #12656) (closes issue #12656) Reported by: dimas - Patches: v3-12656.patch uploaded by dimas (license 88) -- with - some modifications by me - - * include/asterisk/audiohook.h: Oddly enough, all of the contents - of audiohook.h were in there twice. I have removed the second - copy. - -2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c: Call waiting tone occurs too often, because - it's getting serviced by both subchannels. (closes issue #11354) - Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded - by Corydon76 (license 14) - -2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell@digium.com> - - * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones - - use ast_safe_sleep() instead of calling the wait application - (thanks to tilghman for pointing these out!) - - * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 - May 2008) | 7 lines Merge changes from - team/russell/iax2-another-fix-to-the-fix As described in the - following post to the asterisk-dev mailing list, only enforce - destination call numbers when processing an ACK. - http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html - (closes issue #12631) ........ - - * apps/app_milliwatt.c: - Mark app_milliwatt dependent on - res_indications (thanks to jsmith) - fix a typo in a log message - (thanks to qwell) - - * apps/app_milliwatt.c: Change milliwatt to use the proper tone by - default (1004 Hz) instead of 1000 Hz. An option is there to use - 1000 Hz for anyone that might want it. - -2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/lock.h: Define also when not DEBUG_THREADS - - * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c, - channels/chan_agent.c, channels/chan_alsa.c, main/utils.c, - include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging - code that ensures that when we do deadlock avoidance, we don't - lose the information about how a lock was originally acquired. - -2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf@digium.com> - - * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes - issue #11721) (closes issue #12726) Reported by: arkadia Tested - by: murf These changes: 1. revert the changes made via bug 10668; - I should have known that such changes, even tho they made sense - at the time, seemed like an omission, etc, were actually integral - to the CDR system via forkCDR. It makes sense to me now that - forkCDR didn't natively end any CDR's, but rather depended on - natively closing them all at hangup time via traversing and - closing them all, whether locked or not. I still don't completely - understand the benefits of setvar and answer operating on locked - cdrs, but I've seen enough to revert those changes also, and stop - messing up users who depended on that behavior. bug 12726 found - reverting the changes fixed his changes, and after a long review - and working on forkCDR, I can see why. 2. Apply the suggested - enhancements proposed in 10668, but in a completely compatible - way. ForkCDR will behave exactly as before, but now has new - options that will allow some actions to be taken that will - slightly modify the outcome and side-effects of forkCDR. Based on - conversations I've had with various people, these small tweaks - will allow some users to get the behavior they need. For - instance, users executing forkCDR in an AGI script will find the - answer time set, and DISPOSITION set, a situation not covered - when the routines were first written. 3. A small problem in the - cdr serializer would output answer and end times even when they - were not set. This is now fixed. - -2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant@digium.com> - - * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 -- - Fixes a bug in chan_iax that uses send_command to poke a peer - while a channel is unlocked in some cases, and because it can - cause seemingly random failures could be related to some bugs in - the tracker... - -2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an - option to use the source IP address of RTP as the destination IP - address of UDPTL when a specific option is enabled. If the remote - side is properly configured (ports forwarded) then UDPTL will - flow. (closes issue #10417) Reported by: cstadlmann - - * channels/chan_sip.c: Fix an issue where codec preferences were - not set on dialogs that were not authenticated via a user or peer - and allow framing to work without rtpmap in the SDP. (closes - issue #12501) Reported by: slimey - -2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher@digium.com> - - * main/cli.c: When showing an error message for a command, don't - shorten the command output, as it tends to confuse the user (it's - fine for suggesting other commands, however). Reported by: - seanbright (on #asterisk-dev) Fixed by: me - -2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Russell noted to me that in the case that - separate threads use their own addressing system, the fix I made - for issue 12376 does not guarantee uniqueness to the datastores' - uids. Though I know of no system that works this way, I am going - to change this right now to prevent trying to track down some - future bug that may occur and cause untold hours of debugging - time to track down. The change involves using a global counter - which increases with each new chanspy_ds which is created. This - guarantees uniqueness. - -2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: NULL character should terminate only commands - back to the core, not log messages to the console. (closes issue - #12731) Reported by: seanbright Patches: - 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14) - Tested by: seanbright - -2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_voicemail.c: small update to the g() option of - app_voicemail to note that gain changes only work on zap channels - right now. issue #12578 shows it's not clear right now. - -2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Add a unique id to the datastore allocated in - app_chanspy since it is possible that multiple spies may be - listening to the same channel. (closes issue #12376) Reported by: - DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut - (license 60) Tested by: destiny6628 (closes issue #12243) - Reported by: atis - -2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher@digium.com> - - * configs/queues.conf.sample: Add a note that pbx_config.so is - needed for Local channels. (Closes issue #12671) - -2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Realtime flag affects construction in - multiple ways, so consulting whether rtcachefriends was set was - done too soon (needed to be done inside build_peer, not just as a - flag to build_peer). Also, fullcontact needed to be - reconstructed, because realtime separates the embedded ';' into - multiple fields. (closes issue #12722) Reported by: barthpbx - Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 - (license 14) Tested by: barthpbx (Much of the discussion happened - on #asterisk-dev for diagnosing this issue) - -2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: Fix a few things I missed to ensure - zt_chan_conf structure is not modified in mkintf - -2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/utils.h: Add format type checking for recently - de-inlined function - - * doc/cli.txt (added), doc/00README.1st: Add information on using - the Asterisk console, including tab command line completion. - (Closes issue #12681) - -2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h, main/utils.c: Don't declare a function - that takes variable arguments as inline, because it's not valid, - and on some compilers, will emit a warning. - http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes - issue #12289) Reported by: francesco_r Patches by Tilghman, final - patch by me - -2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: Also remove preamble from asynchronous events - (reported by jsmith on #asterisk-dev) - - * funcs/func_realtime.c: Take into account the length of delimiters - when calculating result string length. (closes issue #12696) - Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch - uploaded by adomjan (license 487) - -2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure - is not modified in mkintf. - -2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Apply the autoframing setting to dialogs - that do not get matched against a user or peer. - -2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_spool.c: Revert accidental commit of the last change - - * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the - output also when -rx is not being used (Related to issue #12702) - -2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Don't filter the magic character in the network - verboser. It gets filtered once it reaches the client. (related - to issue #12702, pointed out by tilghman) - - * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose - marker in front of every message from ast_verbose() being sent to - remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose - marker. (related to issue #12702) - - * main/asterisk.c: Don't display the verbose marker for calls to - ast_verbose() that do not include a VERBOSE_PREFIX in front of - the message. (closes issue #12702) Reported by: johnlange Patched - by me - -2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: Pass a pointer for the conf parameter to the - function mkintf rather than the whole zt_chan_conf structure. - -2008-05-20 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.20 released. - -2008-05-14 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.20-rc3 released. - -2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Accept text messages even with Content-Type: - text/plain;charset=Södermanländska - -2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, include/asterisk/lock.h: A change to the way - channel locks are handled when DEBUG_CHANNEL_LOCKS is defined. - After debugging a deadlock, it was noticed that when - DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin - of channel locks is obscured by the fact that all channel locks - appear to happen in the function ast_channel_lock(). This code - change redefines ast_channel_lock to be a macro which maps to - __ast_channel_lock(), which then relays the proper file name, - line number, and function name information to the core lock - functions so that this information will be displayed in the case - that there is some sort of locking error or core show locks is - issued. - -2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell@digium.com> - - * channels/chan_local.c: Fix a deadlock involving channel - autoservice and chan_local that was debugged and fixed by - mmichelson and me. We observed a system that had a bunch of - threads stuck in ast_autoservice_stop(). The reason these threads - were waiting around is because this function waits to ensure that - the channel list in the autoservice thread gets rebuilt before - the stop() function returns. However, the autoservice thread was - also locked, so the autoservice channel list was never getting - rebuilt. The autoservice thread was stuck waiting for the channel - lock on a local channel. However, the local channel was locked by - a thread that was stuck in the autoservice stop function. It - turned out that the issue came down to the local_queue_frame() - function in chan_local. This function assumed that one of the - channels passed in as an argument was locked when called. - However, that was not always the case. There were multiple cases - in which this channel was not locked when the function was - called. We fixed up chan_local to indicate to this function - whether this channel was locked or not. The previous assumption - had caused local_queue_frame() to improperly return with the - channel locked, where it would then never get unlocked. (closes - issue #12584) (related to issue #12603) - - * main/autoservice.c: Fix an issue that I noticed in autoservice - while mmichelson and I were debugging a different problem. I - noticed that it was theoretically possible for two threads to - attempt to start the autoservice thread at the same time. This - change makes the process of starting the autoservice thread, - thread-safe. - -2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp@digium.com> - - * channels/chan_alsa.c: Use the right flag to open the audio in - non-blocking. (closes issue #12616) Reported by: - nicklewisdigiumuser - -2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: If the socket dies (read returns 0=EOF), return - immediately. (Closes issue #12637) - -2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson@digium.com> - - * main/utils.c: If a thread holds no locks, do not print any - information on the thread when issuing a core show locks command. - This will help to de-clutter output somewhat. Russell said it - would be fine to place this improvement in the 1.4 branch, so - that's why it's going here too. - -2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: - Improve res_ninit and res_ndestroy autoconf logic on the Darwin - platform. - -2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Remove debug output. - - * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 - May 2008) | 25 lines Fix a race condition that bbryant just found - while doing some IAX2 testing. He was running Asterisk trunk - running IAX2 calls through a few Asterisk boxes, however, the - audio was extremely choppy. We looked at a packet trace and saw a - storm of INVAL and VNAK frames being sent from one box to - another. It turned out that what had happened was that one box - tried to send a CONTROL frame before the 3 way handshake had - completed. So, that frame did not include the destination call - number, because it didn't have it yet. Part of our recent work - for security issues included an additional check to ensure that - frames that are supposed to include the destination call number - have the correct one. This caused the frame to be rejected with - an INVAL. The frame would get retransmitted for forever, rejected - every time ... This race condition exists in all versions that - got the security changes, in theory. However, it is really only - likely that this would cause a problem in Asterisk trunk. There - was a control frame being sent (SRCUPDATE) at the _very_ - beginning of the call, which does not exist in 1.2 or 1.4. - However, I am fixing all versions that could potentially be - affected by the introduced race condition. These changes are what - bbryant and I came up with to fix the issue. Instead of simply - dropping control frames that get sent before the handshake is - complete, the code attempts to wait a little while, since in most - cases, the handshake will complete very quickly. If it doesn't - complete after yielding for a little while, then the frame gets - dropped. ........ - - * channels/chan_sip.c: Don't give up on attempting an outbound - registration if we receive a 408 Timeout. (closes issue #12323) - - * contrib/scripts/postgres_cdr.sql (removed): remove - postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as - well (closes issue #9676) - - * contrib/init.d/rc.debian.asterisk: Don't exit the script if - Asterisk is not running. (closes issue #12611) - - * main/pbx.c: Don't use a channel before checking for channel - allocation failure. (closes issue #12609) Reported by: edantie - - * contrib/init.d/rc.debian.asterisk: Use the same method for - executing Asterisk as the rest of the script. (closes issue - #12611) Reported by: b_plessis - -2008-05-07 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.20-rc2 released. - -2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Track peer references when stored in the - sip_pvt struct as the peer related to a qualify ping or a - subscription. This fixes some realtime related crashes. (closes - issue #12588) (closes issue #12555) - -2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker@digium.com> - - * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | - 7 lines read requires an argument on some non-bash shells (closes - issue #12593) Reported by: bkruse Patches: - getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) - ........ - - * res/res_musiconhold.c: Switch to using ast_random() rather than - just rand(). This does not fix the bug reported, but I believe it - is correct. (from issue #12446) Patches: bug_12446.diff uploaded - by snuffy (license 35) - -2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: Don't print the terminating NUL. (Closes issue - #12589) - -2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Add in missing argument. - -2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, main/logger.c: Separate verbose output from CLI - output, by using a preamble. (closes issue #12402) Reported by: - Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt - uploaded by Corydon76 (license 14) - 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by - Corydon76 (license 14) - -2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp@digium.com> - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, codecs/codec_speex.c, - configure.ac: Make sure that either the main speex library - contains preprocess functions or that speexdsp does. If both fail - then speex stuff can not be built. - -2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Don't consider a caller "handled" until the - caller is bridged with a queue member. There was too much of an - opportunity for the member to hang up (either during a delay, - announcement, or overly long agi) between the time that he - answered the phone and the time when he actually was bridged with - the caller. The consequence of this was that if the member hung - up in that interval, then proper abandonment details would not be - noted in the queue log if the caller were to hang up at any point - after the member hangup. (closes issue #12561) Reported by: - ablackthorn - -2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher@digium.com> - - * Makefile: Reverse order, such that user configs override default - selections - - * include/asterisk/res_odbc.h: Err, the documentation on the return - value of ast_odbc_backslash_is_escape is exactly backwards. - -2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Avoid putting opaque="" in Digest - authentication. This patch came from switchvox. It fixes - authentication with Primus in Canada, and has been in use for a - very long time without causing problems with any other providers. - (closes issue AST-36) - -2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, - 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.slackware.asterisk: When starting Asterisk, bug - out if Asterisk is already running. (closes issue #12525) - Reported by: explidous Patches: 20080428__bug12525.diff.txt - uploaded by Corydon76 (license 14) Tested by: mvanbaak - -2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp@digium.com> - - * configure, acinclude.m4: Expand the test function for GCC - attributes so that more complex attributes are properly - recognized. - - * include/asterisk/compiler.h: For my next trick I will make these - work with what our autoconf header file gives us. - - * configure, acinclude.m4: Treat warnings as errors when checking - if a GCC attribute exists. We have to do this as GCC will just - ignore the attribute and pop up a warning, it won't actually fail - to compile. - -2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant@digium.com> - - * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, - configure.ac, CHANGES: Add new "pri show version" command to show - the libpri version for support reasons. - -2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/sched.h: Clarify a comment that was, well, just - wrong. It turns out that ignoring the way that macros expand. - Instead, I have clarified in the comment why the macro will work - even if the scheduler id for the task to be deleted changes - during the execution of the macro. - -2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h: Change the comment of deprecated to - an actual compiler deprecation - - * main/utils.c: '#' is another reserved character for URIs that - also needs to be escaped. (closes issue #10543) Reported by: - blitzrage Patches: 20080418__bug10543.diff.txt uploaded by - Corydon76 (license 14) - -2008-05-01 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.20-rc1 released. - -2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell@digium.com> - - * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c: - Merge changes from team/russell/iax2_find_callno and - iax2_find_callno_1.4 These changes address a critical performance - issue introduced in the latest release. The fix for the latest - security issue included a change that made Asterisk randomly - choose call numbers to make them more difficult to guess by - attackers. However, due to some inefficient (this is by far, an - understatement) code, when Asterisk chose high call numbers, - chan_iax2 became unusable after just a small number of calls. On - a small embedded platform, it would not be able to handle a - single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run - more than about 16 IAX2 channels. Ouch. These changes address - some performance issues of the find_callno() function that have - bothered me for a very long time. On every incoming media frame, - it iterated through every possible call number trying to find a - matching active call. This involved a mutex lock and unlock for - each call number checked. So, if the random call number chosen - was 20000, then every media frame would cause 20000 locks and - unlocks. Previously, this problem was not as obvious since - Asterisk always chose the lowest call number it could. A second - container for IAX2 pvt structs has been added. It is an astobj2 - hash table. When we know the remote side's call number, the pvt - goes into the hash table with a hash value of the remote side's - call number. Then, lookups for incoming media frames are a very - fast hash lookup instead of an absolutely insane array traversal. - In a quick test, I was able to get more than 3600% more IAX2 - channels on my machine with these changes. - -2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created - in Huntsville together with Mark M (putnopvut) (closes issue - #12363) Reported by: jvandal Tested by: putnopvut, oej - -2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming@digium.com> - - * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro - for indexing through the iaxs/iaxsl arrays so that the size of - the arrays can be adjusted in one place, and change the size of - the arrays from 32768 calls to 2048 calls when LOW_MEMORY is - defined - - * Makefile.rules: pay attention to *all* header files for - dependency tracking, not just the local ones (inspired by r578 of - asterisk-addons by tilghman) - -2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel - variables instead of the channel's macrocontext and macroexten - fields. This is needed because if macros are daisy-chained, the - incorrect context and extension are placed on the new channel. I - also added locking to the channel prior to accessing these - variables as noted in trunk's janitor project file. (closes issue - #12549) Reported by: darren1713 Patches: - app_queue.c.macroextenpatch uploaded by darren1713 (license 116) - (with modifications from me) Tested by: putnopvut - -2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker@digium.com> - - * res/res_config_pgsql.c: Change warning message to debug, since - there are cases where 0 results is perfectly fine. - -2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming@digium.com> - - * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr - 2008) | 2 lines stop script from appending source code if run - multiple times ........ - -2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are - embedded, they take on a different name, without the ".so" - extension. Specifically check for this name, when we're checking - if a module is loaded. (Closes issue #12534) - -2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright@gmail.com> - - * configure, configure.ac: When we don't explicitly pass a path to - the --with-tds configure option, we may end up finding tds.h in - /usr/local/include instead of /usr/include. If this happens, the - grep that looks for the version (from tdsver.h) will fail and - we'll have some problems during the build. - -2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/vmail.cgi: Clicking forward without selecting a - message leaves an errant .lock file. (closes issue #12528) - Reported by: pukepail Patches: patch.diff uploaded by pukepail - (license 431) - -2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Use consistent logic for checking to see if - a call number has been chosen yet. Also, remove some redundant - logic I recently added in a fix. - -2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Move the unlock of the spyee channel to - outside the start_spying() function so that the channel is not - unlocked twice when using whisper mode. - -2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher@digium.com> - - * configs/zapata.conf.sample, configs/iax.conf.sample, - configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference - documentation files that actually exist. (closes issue #12516) - Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded - by linuxmaniac (license 472) - -2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Re-invite RTP during a masquerade so that, - for instance, an AMI redirect of two channels which are natively - bridged will preserve audio on both channels. This prevents a - problem with Asterisk not re-inviting due to one of the channels - having being a zombie. (closes issue #12513) Reported by: - mneuhauser Patches: - asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by - mneuhauser (license 425) - - * apps/app_queue.c: Output of channel variables when - eventwhencalled=vars was set was being truncated two characters. - This patch corrects the problem. (closes issue #12493) Reported - by: davidw - - * channels/chan_local.c: Resolve a deadlock in chan_local by - releasing the channel lock temporarily. (closes issue #11712) - Reported by: callguy Patches: 11712.patch uploaded by putnopvut - (license 60) Tested by: acunningham - -2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c: Ensure that when we set the accountcode, - it actually shows up in the CDR. (Fix for AMI Originate) (Closes - issue #12007) - -2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a silly mistake in a change I made - yesterday that caused chan_iax2 to blow up very quickly. (issue - #12515) - -2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Only have one max-forwards header in - outbound REFERs. Discovered in the Asterisk SIP Masterclass in - Orlando. Thanks Joe! - -2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell@digium.com> - - * main/http.c: Improve some broken cookie parsing code. Previously, - manager login over HTTP would only work if the mansession_id - cookie was first. Now, the code builds a list of all of the - cookies in the Cookie header. This fixes a problem observed by - users of the Asterisk GUI. (closes AST-20) - - * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting - the correct next channel to not always work. Also, remove setting - the amount of time to wait for a digit from 5 seconds back down - to 1/10 of a second. I believe this was so the beep didn't get - played over and over really fast, but a while back I put in - another fix for that issue. (closes issue #12498) Reported by: - jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by - jsmith (license 15) - -2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker@digium.com> - - * res/res_musiconhold.c: Fix reload/unload for res_musiconhold - module. (closes issue #11575) Reported by: sunder Patches: - M11575_14_rev3.diff uploaded by junky (license 177) - bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176) - -2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell@digium.com> - - * main/manager.c, include/asterisk/manager.h: Store the manager - session ID explicitly as 4 byte ID instead of a ulong. The - mansession_id cookie is coded to be limited to 8 characters of - hex, and this could break logins from 64-bit machines in some - cases. (inspired by AST-20) - - * channels/chan_iax2.c: Fix find_callno_locked() to actually return - the callno locked in some more cases. - -2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add 502 support for both directions, not - only one... (see r114571) - -2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: Instead of stopping dialplan execution when SayNumber - attempts to say a large number that it can not print out a - message informing the user and continue on. (closes issue #12502) - Reported by: bcnit - -2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Treat a 502 just like a 503, when it comes - to processing a response code - -2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: When we receive a full frame that is - supposed to contain our call number, ensure that it has the - correct one. (closes issue #10078) (AST-2008-006) - - * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I - thought I was going to be able to leave 1.4 alone, but that was - not the case. I ran into some problems with G.722 in 1.4, so I - have merged in all of the fixes in this area that I have made in - trunk/1.6.0, and things are happy again. - - * res/res_musiconhold.c: Trivial change to read the number of - samples from a frame before calling ast_write() - - * res/res_features.c: After a parked call times out, allow the call - back to the parker to time out. (closes issue #10890) - - * channels/chan_iax2.c: If the dial string passed to the call - channel callback does not indicate an extension, then consider - the extension on the channel before falling back to the default. - (closes issue #12479) Reported by: darren1713 Patches: - exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license - 116) - - * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from - team/russell/issue_9520 These changes make sure that the - reference count for sip_peer objects properly reflects the fact - that the peer is sitting in the scheduler for a scheduled - callback for qualifying peers or for expiring registrations. - Without this, it was possible for these callbacks to happen at - the same time that the peer was being destroyed. This was - especially likely to happen with realtime peers, and for people - making use of the realtime prune CLI command. (closes issue - #9520) Reported by: kryptolus Committed patch by me - -2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only drop audio if we receive it without a - progress indication. We allow other frames through such as DTMF - because they may be needed to complete the call. (closes issue - #12440) Reported by: aragon - -2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher@digium.com> - - * apps/app_playback.c: Ensure that help text terminates with a - newline - - * res/res_musiconhold.c: MOH usage information needs a terminating - newline, or else "asterisk -rx 'help moh reload'" will hang. - Reported via -dev list, fixed by me. - -2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell@digium.com> - - * main/manager.c: Don't destroy a manager session if poll() returns - an error of EAGAIN. - - * Makefile: ensure directories are created before we try to install - stuff into them - - * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for - bininstall - -2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c, main/callerid.c: Clearing up error messages - so they make a bit more sense. Also removing a redundant error - message. Issue AST-15 - -2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL) - (closes issue #12476) Reported by: davidw Patch by me - -2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright@gmail.com> - - * channels/chan_sip.c: Only complete the SIP channel name once for - 'sip show channel <channel>' - -2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher@digium.com> - - * apps/app_setcallerid.c: For consistency sake, ensure that the - values that ${CALLINGPRES} returns are valid as an input to - SetCallingPres. (Closes issue #12472) - -2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell@digium.com> - - * main/autoservice.c: Remove redundant safety net. The check for - the autoservice channel list state accomplishes the same goal in - a better way. (issue #12470) Reported By: atis - -2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Declaration of the peer channel in this scope - was making it so the peer variable defined in the outer scope was - never set properly, therefore making iterating through the - channel list always restart from the beginning. This bug would - have affected anyone who called chanspy without specifying a - first argument. (closes issue #12461) Reported by: stever28 - - * main/frame.c, include/asterisk/dsp.h: Add prototype for - ast_dsp_frame_freed. I'm not sure how this was compiling - before... - - * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was - possible for a reference to a frame which was part of a freed DSP - to still be referenced, leading to memory corruption and eventual - crashes. This code change ensures that the dsp is freed when we - are finished with the frame. This change is very similar to a - change Russell made with translators back a month or so ago. - (closes issue #11999) Reported by: destiny6628 Patches: - 11999.patch uploaded by putnopvut (license 60) Tested by: - destiny6628, victoryure - -2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell@digium.com> - - * Makefile: Fix the bininstall target to install from subdirs, as - well. (closes issue AST-8, patch from bmd at switchvox) - -2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Use keepalives effectively in order diagnose - bug #12432. - -2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Add special case for when the agi cannot be - executed, to comply with the documentation that we return failure - in that case. (closes issue #12462) Reported by: fmueller - Patches: 20080416__bug12462.diff.txt uploaded by Corydon76 - (license 14) Tested by: fmueller - -2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright@gmail.com> - - * apps/app_chanspy.c: Make sure we have enough room for the - recording's filename. - -2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by - initializing the structure to all zeroes in case it contains - fields that we don't write values into (which it does as of - Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian - -2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_vpb.cc: Backport revisions for latest vpb drivers - to 1.4 (Closes issue #12457) - -2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Fix "fallthrough" behavior here, so config - options in a previously configured user don't override settings - in general. (closes issue #12458) Reported by: tzafrir Patches: - chanzap_users_sections.diff uploaded by tzafrir (license 46) - -2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Include the proper headers for using mkdir on - FreeBSD. (closes issue #12430) Reported by: ys Patches: - app_meetme.c.diff uploaded by ys (license 281) - -2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Handle subscribe queues in all situations... - Thanks to festr_ on irc for telling me about this bug. - -2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker@digium.com> - - * contrib/scripts/autosupport: Update Digium autosupport script, - for more useful information. (closes issue #12452) Reported by: - angler Patches: autosupport.diff uploaded by angler (license 106) - - * apps/app_queue.c: Allow autofill to work in the general section - of queues.conf. Additionally, don't try to (re)set options when - they have empty values in realtime (all unset columns would have - an empty value). (closes issue #12445) Reported by: atis Patches: - 12445-autofill.diff uploaded by qwell (license 4) - - * channels/chan_h323.c: The call_token on the pvt can occasionally - be NULL, causing a crash. If it is NULL, we can skip this - channel, since it can't the one we're looking for. (closes issue - #9299) Reported by: vazir - -2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: Increase the retry count when attempting to show - channels. This apparently cleared an issue someone was seeing - when attempting to show channels when the load was high. (closes - issue #11667) Reported by: falves11 Patches: 11677.txt uploaded - by russell (license 2) Tested by: falves11 - - * apps/app_dial.c, apps/app_queue.c: If the datastore has been - moved to another channel due to a masquerade, then freeing the - datastore here causes an eventual double free when the new - channel hangs up. We should only free the datastore if we were - able to successfully remove it from the channel we are - referencing (i.e. the datastore was not moved). (closes issue - #12359) Reported by: pguido - - * main/channel.c: Save a local copy of the generate callback prior - to unlocking the channel in case the generate callback goes NULL - on us after the channel is unlocked. Thanks to Russell for - pointing this need out to me. - -2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: It is possible for the remote side to say - they want T38 but not give any capabilities. (closes issue - #12414) Reported by: MVF - - * main/rtp.c: Don't change the SSRC when a new source comes into - play, this might happen quite often and depending on the remote - side... they might not like this. (closes issue #12353) Reported - by: dimas - -2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson@digium.com> - - * channels/chan_iax2.c: Several places in the code called - find_callno() (which releases the lock on the pvt structure) and - then immediately locked the call and did things with it. - Unfortunately, the call can disappear between the find_callno and - the lock, causing Bad Stuff(tm) to happen. Added - find_callno_locked() function to return the callno withtout - unlocking for instances that it is needed. (issue #12400) - Reported by: ztel - -2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker@digium.com> - - * main/pbx.c: It's possible that a channel can have an async goto - on the successful execution of an application as well. Closes - issue #12172. - -2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson@digium.com> - - * res/res_features.c: Fix a race condition that may happen between - a sip hangup and a "core show channel" command. This patch adds - locking to prevent the resulting crash. (closes issue #12155) - Reported by: tsearle Patches: show_channels_crash2.patch uploaded - by tsearle (license 373) Tested by: tsearle - - * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when - LOW_MEMORY is enabled. - - * channels/chan_sip.c: Be sure that we're not about to set - bridgepvt NULL prior to dereferencing it. (closes issue #11775) - Reported by: fujin - -2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker@digium.com> - - * main/file.c: Only try to prefix language if we are not using an - absolute path (suffix it otherwise). - en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes - issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff - uploaded by qwell (license 4) Tested by: kuj, qwell - -2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Forgot the 1.4 branch for russian language - fix. (closes issue #12404) Reported by: IgorG Patches: - voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20) - - * apps/app_meetme.c: Create the directory where name recordings - will go if it does not exist. (closes issue #12311) Reported by: - rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) - - * channels/chan_sip.c: Don't add custom URI options if they don't - exist OR they are empty. (closes issue #12407) Reported by: - homesick Patches: uri_options-1.4.diff uploaded by homesick - (license 91) - -2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: We need to set the persistant_route [sic] - parameter for the sip_pvt during the initial INVITE, no matter if - we're building the route set from an INVITE request or response. - (closes issue #12391) Reported by: benjaminbohlmann Tested by: - benjaminbohlmann - -2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does - not exist in cdr.conf, then an unload/load sequence is needed to - correct the problem. Track whether the load succeeded with a - variable, so we can fix this with a simple reload event, instead. - -2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: If we receive an AUTHREQ from the remote - server and we are unable to reply (for example they have a secret - configured, but we do not) then queue a hangup frame on the - Asterisk channel. This will cause the channel to hangup and a - HANGUP to be sent via IAX2 to the remote side which is the proper - thing to do in this scenario. (closes issue #12385) Reported by: - viraptor - -2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not - a reinvite), then we should not send a BYE. (closes issue #12392) - Reported by: fnordian Patches: chan_sip.patch uploaded by - fnordian (license 110) with small modification from me - -2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson@digium.com> - - * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind - happy - -2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson@digium.com> - - * apps/app_parkandannounce.c: Fix potential buffer overflow that - could happen if more than 100 announce files were specified when - calling ParkAndAnnounce. This overflow is not exploitable - remotely and so there is no need for a security advisory. (closes - issue #12386) Reported by: davidw - -2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Add a little more that is required for - previously added devices. - - * channels/chan_skinny.c: Add support for several new(ish) devices - - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver - for providing me the required information. - - * main/asterisk.c: Work around some silliness caused by - sys/capability.h - this should fix compile errors a number of - users have been experiencing. - -2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/astgenkey.8: Add security note on astgenkey's - manpage. (closes issue #12373) Reported by: lmamane Patches: - 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14) - - * channels/chan_sip.c: Move check for still-bridged channels out a - little further, to avoid possible deadlocks. (Closes issue - #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt - uploaded by Corydon76 (license 14) Tested by: callguy - -2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp@digium.com> - - * include/asterisk/slinfactory.h, main/slinfactory.c, - main/audiohook.c: If audio suddenly gets fed into one side of a - channel after a lapse of frames flush the other factory so that - old audio does not remain in the factory causing the sync code to - not execute. (closes issue #12296) Reported by: jvandal - -2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This - fixes a for loop (in realtime_peer) to check all the - ast_variables the loop was intending to test rather than just the - first one. The change exposed the problem of calling memcpy on a - NULL pointer, in this case the passed in sockaddr_in struct which - is now checked. - -2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Allow - playback with noanswer (and add earlyrtp option). (closes issue - #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn - (license 30) Tested by: pj, qwell, DEA, wedhorn - -2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_strings.c: Force ast_mktime() to check for DST, since - strptime(3) does not. (Closes issue #12374) - -2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: This fix prevents a deadlock that was experienced - in chan_local. There was deadlock prevention in place in - chan_local, but it would not work in a specific case because the - channel was recursively locked. By unlocking the channel prior to - calling the generator's generate callback in - ast_read_generator_actions(), we prevent the recursive locking, - and therefore the deadlock. (closes issue #12307) Reported by: - callguy Patches: 12307.patch uploaded by putnopvut (license 60) - Tested by: callguy - -2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_sip.c: (closes issue #12362) (closes issue #12372) - Reported by: vinsik Tested by: tecnoxarxa This one line change - makes an if inside a for loop (in realtime_peer) check all the - ast_variables the loop was intending to test rather than just the - first one. - -2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Free newly allocated channel before - returning - - * channels/chan_gtalk.c: Prevent call connections when codecs don't - match. (closes issue #10604) Reported by: keepitcool Patches: - branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested - by: phsultan - -2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp@digium.com> - - * main/Makefile: Pass in the path to Zaptel for systems that - install Zaptel headers in a separate location. - - * main/asterisk.c: One thing at a time... let's get 1.4 building. - -2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/asterisk.c: add a Zaptel timer check to verify the timer is - responding when Zaptel support is compiled into Asterisk and - Zaptel drivers are loaded. This will help people not waste their - valuable time debugging side effects. - -2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c: Fix the testing of the "res" variable so - that it is more logically correct and makes the correct warning - and debug messages print. (closes issue #12361) Reported by: - one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 - (license 23) - - * main/manager.c: Fix a race condition in the manager. It is - possible that a new manager event could be appended during a - brief time when the manager is not waiting for input. If an event - comes during this period, we need to set an indicator that there - is an event pending so that the manager doesn't attempt to wait - forever for an event that already happened. (closes issue #12354) - Reported by: bamby Patches: manager_race_condition.diff uploaded - by bamby (license 430) (comments added by me) - - * apps/app_queue.c: Ensure that there is no timeout if none is - specified. (closes issue #12349) Reported by: johnlange - -2008-04-01 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.19 released. - -2008-03-28 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.19-rc4 released. - -2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker@digium.com> - - * formats/format_wav_gsm.c: The file size of WAV49 does not need to - be an even number. (closes issue #12128) Reported by: mdu113 - Patches: 12128-noevenlength.diff uploaded by qwell (license 4) - Tested by: qwell, mdu113 - -2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher@digium.com> - - * doc/valgrind.txt: Update debugging text, since Valgrind - eliminated the --log-file-exactly option. (Closes issue #12320) - - * main/acl.c: For FreeBSD, at least, the ifa_addr element could be - NULL. (closes issue #12300) Reported by: festr Patches: - acl.c.patch uploaded by festr (license 443) - -2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf@digium.com> - - * apps/app_playback.c, main/pbx.c: These small documentation - updates made in response to a query in asterisk-users, where a - user was using Playback, but needed the features of Background, - and had no idea that Background existed, or that it might provide - the features he needed. I thought the best way to avert these - kinds of queries was to provide "See Also" references in all - three of "Background", "Playback", "WaitExten". Perhaps a project - to do this with all related apps is in order. - - * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue - #12302) Reported by: pj Tested by: murf These changes will set a - channel variable ~~EXTEN~~ just before generating code for a - switch, with the value of ${EXTEN}. The exten is marked as having - a switch, and ever after that, till the end of the exten, we - substitute any ${EXTEN} with ${~~EXTEN~~} instead in application - arguments; (and the ${EXTEN: also). The reason for this, is that - because switches are coded using separate extensions to provide - pattern matching, and jumping to/from these switch extensions - messes up the ${EXTEN} value, which blows the minds of users. - -2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker@digium.com> - - * main/frame.c: Put this flag back so we don't change the API. - - * main/frame.c: Remove excessive smoother optimization that was - causing audio glitches (small "pops") after (about 200ms later) - an "incorrectly" sized frame was received. While it would be very - nice to keep this as optimized as possible, it makes no sense for - the smoother to be dropping random bits of audio like this. Isn't - that the whole point of a smoother? Closes issue #12093. - -2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp@digium.com> - - * contrib/scripts/autosupport: Update autosupport script. (closes - issue #12310) Reported by: angler Patches: autosupport.diff - uploaded by angler (license 106) - -2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming@digium.com> - - * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar - 2008) | 2 lines update UPGRADE notes to document usage of the - script ........ - -2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: This code change is made just for - clarification. It does exactly the same thing as before. It just - doesn't look as wrong. - - * apps/app_voicemail.c: Add a lock to the vm_state structure and - use the lock around mail_open calls to prevent concurrent access - of the same mailstream. This, along with trunk's ability to - configure TCP timeouts for IMAP storage will help to prevent - crashes and hangs when using voicemail with IMAP storage. (closes - issue #10487) Reported by: ewilhelmsen - -2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): - Merged revisions 111019 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar - 2008) | 2 lines add a script to make getting the iLBC source code - simple for end users ........ - -2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: If we are requested to authenticate a - reinvite make sure that it contains T38 SDP if need be. (closes - issue #11995) Reported by: fall - - * channels/chan_iax2.c: Make sure that full video frames are sent - whenever the 15 bit timestamp rolls over. (closes issue #11923) - Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by - mihai (license 94) - -2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming@digium.com> - - * UPGRADE.txt: add note that the user will need to enable - codec_ilbc to get it to build - - * codecs/ilbc/StateConstructW.h (removed), - codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h - (removed), codecs/ilbc/getCBvec.c (removed), - codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c - (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c - (removed), codecs/ilbc/getCBvec.h (removed), - codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h - (removed), codecs/ilbc/FrameClassify.c (removed), - codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed), - codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h - (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c - (removed), codecs/ilbc/anaFilter.c (removed), - codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c - (removed), codecs/ilbc/doCPLC.h (removed), - codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, - codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c - (removed), codecs/ilbc/createCB.h (removed), CHANGES, - codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h - (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed), - codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), - codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c - (removed), codecs/ilbc/iCBSearch.h (removed), - codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), - codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c - (removed), codecs/ilbc/hpOutput.h (removed), - codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, - codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h - (removed), codecs/ilbc/iCBConstruct.c (removed), - codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h - (removed), codecs/ilbc/syntFilter.h (removed), - codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c - (removed): Merged revisions 110869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar - 2008) | 2 lines due to licensing restrictions, we cannot - distribute the source code for iLBC encoding and decoding... so - remove it, and add instructions on how the user can obtain it - themselves ........ - -2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker@digium.com> - - * cdr/cdr_custom.c: Make file access in cdr_custom similar to - cdr_csv. Fixes issue #12268. Patch borrowed from r82344 - -2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_sip.c: This one line change makes an if inside a - for loop (in realtime_peer) check all the ast_variables the loop - was intending to test rather than just the first one. - -2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: When reverting a commit, I accidentally left - in this bit which was an experiment to see what would happen. It - passed the compile test, and I didn't notice I had left this - change in too. So this is a revert of a revert...sort of. - -2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp@digium.com> - - * include/asterisk/options.h, main/asterisk.c, Makefile, - main/app.c: Add an option (transmit_silence) which transmits - silence during both Record() and DTMF generation. The reason this - is an option is that in order to transmit silence we have to - setup a translation path. This may not be needed/wanted in all - cases. (closes issue #10058) Reported by: tracinet - -2008-03-24 19:17 +0000 [r110618] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: This is a revert for revision 108288. The - reason is that that revision was not for an actual bug fix per - se, and so it really should not have been in 1.4 in the first - place. Plus, people who compile with DO_CRASH are more likely to - encounter a crash due to this change. While I think the usage of - DO_CRASH in ast_sched_del is a bit absurd, this sort of change is - beyond the scope of 1.4 and should be done instead in a developer - branch based on trunk so that all scheduler functions are fixed - at once. I also am reverting the change to trunk and 1.6 since - they also suffer from the DO_CRASH potential. (closes issue - #12272) Reported by: qq12345 - -2008-03-24 17:34 +0000 [r110614] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message. - -2008-03-21 14:32 +0000 [r110474] Jason Parker <jparker@digium.com> - - * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on - mips platforms either. (closes issue #12270) Reported by: - zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt - (license 33) - -2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant <russell@digium.com> - - * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms - to 50 ms in the autoservice thread. This really should not make a - difference except in very rare cases. That case would be that all - of the channels in autoservice are not generating any frames. In - that case, this change reduces the potential amount of time that - a thread waits in ast_autoservice_stop() for the autoservice - thread to wrap back around to the beginning of its loop. (closes - issue #12266, reported by dimas) - - * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions - 110335 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) - | 6 lines Fix some very broken code that was introduced in 1.2.26 - as a part of the security fix. The dnsmgr is not appropriate - here. The dnsmgr takes a pointer to an address structure that a - background thread continuously updates. However, in these cases, - a stack variable was passed. That means that the dnsmgr thread - would be continuously writing to bogus memory. ........ - - * apps/app_meetme.c: Fix a bug where when calls on the trunk side - hang up while on hold, the state is not properly reflected. - (closes issue #11990, reported by anakaoka, patched by me) - -2008-03-19 20:33 +0000 [r110083] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Add a missing unlock in the case that memory - allocation fails in app_chanspy. Thanks to Russell for confirming - that this was an issue. - -2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Add sanity checking for position resuming. - We *have* to make sure that the position does not exceed the - total number of files present, and we have to make sure that the - position's filename is the same as previous. These values can - change if a music class is reloaded and give unpredictable - behavior. (closes issue #11663) Reported by: junky - - * main/rtp.c: Make sure that the mark bit does not incorrectly - cause video frame timestamps to be calculated as if they are - audio frames. (closes issue #11429) Reported by: sperreault - Patches: 11429-frametype.diff uploaded by qwell (license 4) - -2008-03-19 17:12 +0000 [r109973] Jason Parker <jparker@digium.com> - - * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml - (added): People report bugs about Asterisk crashing with DO_CRASH - enabled was getting a little silly... Now we only show certain - cflags when you run configure with --enable-dev-mode - (corresponding menuselect change to follow) - -2008-03-19 15:41 +0000 [r109908] Steve Murphy <murf@digium.com> - - * main/config.c: (closes issue #11442) Reported by: tzafrir - Patches: 11442.patch uploaded by murf (license 17) Tested by: - murf I didn't give tzafrir very much time to test this, but if he - does still have remaining issues, he is welcome to re-open this - bug, and we'll do what is called for. I reproduced the problem, - and tested the fix, so I hope I am not jumping by just going - ahead and committing the fix. The problem was with what file_save - does with templates; firstly, it tended to print out multiple - options: [my_category](!)(templateref) instead of - [my_category](!,templateref) which is fixed by this patch. - Nextly, the code to suppress output of duplicate declarations - that would occur because the reader copies inherited declarations - down the hierarchy, was not working. Thus: [master-template](!) - mastervar = bar [template](!,master-template) tvar = value - [cat](template) catvar = val would be rewritten as: ;! ;! - Automatically generated configuration file ;! Filename: - experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: - Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! - [master-template](!) mastervar = bar - [template](!,master-template) mastervar = bar tvar = value - [cat](template) mastervar = bar tvar = value catvar = val This - has been fixed. Since the config reader 'explodes' inherited vars - into the category, users may, in certain circumstances, see - output different from what they originally entered, but it should - be both correct and equivalent. - -2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant <russell@digium.com> - - * main/utils.c: Tweak spacing in a recent change because I'm very - picky. - - * apps/app_chanspy.c: Fix one place where the chanspy datastore - isn't removed from a channel. (issue #12243, reported by atis, - patch by me) - -2008-03-18 20:52 +0000 [r109713] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: This patch makes it so that all queue member - status changes are handled through device state code. This - removes several problems people were seeing where their queue - members would get into an "unknown" state. Huge props go to atis - on this one since he was the one who found the code section that - was causing the problem and proposed the solution. I just wrote - what he suggested :) (closes issue #12127) Reported by: atis - Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested - by: atis, jvandal - -2008-03-18 19:23 +0000 [r109648] Jason Parker <jparker@digium.com> - - * codecs/log2comp.h: Allow codecs that use log2comp (g726) to - compile correctly on x86 with gcc4 optimizations. (closes issue - #12253) Reported by: fossil Patches: log2comp.patch uploaded by - fossil (license 140) - -2008-03-18 17:58 +0000 [r109575] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Make sure an agent doesn't try to send - dtmf to a NULL channel closes issue #12242 Reported by Yourname - -2008-03-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.19-rc3 released. - -2008-03-18 16:25 +0000 [r109482] Terry Wilson <twilson@digium.com> - - * include/asterisk/astobj.h: Fix character string being treated ad - format string - -2008-03-18 15:10 +0000 [r109393] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) | - 3 lines Do not return with a successful authentication if the - From header ends up empty. (AST-2008-003) ........ - -2008-03-18 14:58 +0000 [r109386] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the - number of payloads accepted, and also make sure a given payload - does not exceed our maximum value. (AST-2008-002) - -2008-03-18 06:37 +0000 [r109309] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ael-ntest23 (added), - pbx/ael/ael-test/ael-ntest23/t1/a.ael (added), - pbx/ael/ael-test/ael-ntest23/t1/b.ael (added), - pbx/ael/ael-test/ael-ntest23/t1/c.ael (added), - pbx/ael/ael-test/ael-ntest23/t2/d.ael (added), - pbx/ael/ael-test/ael-ntest23/t2/e.ael (added), - pbx/ael/ael-test/ael-ntest23/t2/f.ael (added), - pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c, - pbx/ael/ael-test/ael-ntest23/t3/g.ael (added), - pbx/ael/ael-test/ael-ntest23/t3/h.ael (added), - pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex, - pbx/ael/ael-test/ael-ntest23/t3/j.ael (added), - pbx/ael/ael-test/ael-ntest23/qq.ael (added), - pbx/ael/ael-test/ael-ntest23/t1 (added), - pbx/ael/ael-test/ael-ntest23/t2 (added), - pbx/ael/ael-test/ael-ntest23/t3 (added), - pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes - issue #11903) Reported by: atis Many thanks to atis for spotting - this problem and reporting it. The fix was to straighten out how - items are placed on and removed from the file stack. Regressions - as well as the provided test case helped to straighten out all - code paths. valgrind was used to make sure all memory allocated - was freed. Sorry for not solving this earlier. I got distracted. - Added the ntest23 regression test, which is mainly a copy of - ntest22, but with a few juicy errors thrown in, to replicate the - kind of error that atis spotted. - -2008-03-17 22:05 +0000 [r109226] Mark Michelson <mmichelson@digium.com> - - * main/utils.c: Fix a logic flaw in the code that stores lock info - which is displayed via the "core show locks" command. The idea - behind this section of code was to remove the previous lock from - the list if it was a trylock that had failed. Unfortunately, - instead of checking the status of the previous lock, we were - referencing the index immediately following the previous lock in - the lock_info->locks array. The result of this problem, under the - right circumstances, was that the lock which we currently in the - process of attempting to acquire could "overwrite" the previous - lock which was acquired. While this does not in any way affect - typical operation, it *could* lead to misleading "core show - locks" output. - -2008-03-17 17:55 +0000 [r109171] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: Update the directory of placed calls on - skinny phones when dialing a channel that does not provide - progress (analog ZAP lines) The phone does handle the double - update on calls to channels that do provide progress and wont - insert duplicate items (closes issue #12239) Reported by: DEA - Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) - -2008-03-17 16:24 +0000 [r109107] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 200 OKs in response to a reinvite need to be - sent reliably. If the remote side does not receive one the dialog - will be torn down. (closes issue #12208) Reported by: atrash - -2008-03-17 15:15 +0000 [r109057] Jason Parker <jparker@digium.com> - - * main/file.c: Backport revision 106439 from trunk. I didn't - realize this was broken in 1.4 as well. Closes issue #12222. - -2008-03-17 14:18 +0000 [r109012] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: Make sure that we release the lock on the - spyee channel if the spyee or spy has hung up (closes issue - #12232) Reported by: atis - -2008-03-16 21:47 +0000 [r108961] Michiel van Baak <michiel@vanbaak.info> - - * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE - (closes issue #12228) Reported by: andrew Patches: SRC.patch - uploaded by andrew (license 240) - -2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant <russell@digium.com> - - * channels/chan_oss.c: Fix a channel name issue. chan_oss registers - the "Console" channel type, but it created channels with an "OSS" - prefix. (closes issue #12194, reported by davidw, patched by me) - - * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to - start networking before asterisk, as well. (closes issue #12200, - reported by and change suggested by reinerotto) - -2008-03-14 16:44 +0000 [r108737] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix a race condition in the SIP packet - scheduler which could cause a crash. chan_sip uses the scheduler - API in order to schedule retransmission of reliable packets (such - as INVITES). If a retransmission of a packet is occurring, then - the packet is removed from the scheduler and retrans_pkt is - called. Meanwhile, if a response is received from the packet as - previously transmitted, then when we ACK the response, we will - remove the packet from the scheduler and free the packet. The - problem is that both the ACK function and retrans_pkt attempt to - acquire the same lock at the beginning of the function call. This - means that if the ACK function acquires the lock first, then it - will free the packet which retrans_pkt is about to read from and - write to. The result is a crash. The solution: 1. If the ACK - function fails to remove the packet from the scheduler and the - retransmit id of the packet is not -1 (meaning that we have not - reached the maximum number of retransmissions) then release the - lock and yield so that retrans_pkt may acquire the lock and - operate. 2. Make absolutely certain that the ACK function does - not recursively lock the lock in question. If it does, then - releasing the lock will do no good, since retrans_pkt will still - be unable to acquire the lock. (closes issue #12098) Reported by: - wegbert (closes issue #12089) Reported by: PTorres Patches: - 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested - by: jvandal - -2008-03-14 14:29 +0000 [r108682] Jason Parker <jparker@digium.com> - - * res/res_musiconhold.c: Fix a potential segfault if chan (or - chan->music_state) is NULL. Closes issue #12210, credit to - edantie for pointing this out. - -2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h: - Fix another issue that was causing crashes in chanspy. This - introduces a new datastore callback, called chan_fixup(). The - concept is exactly like the fixup callback that is used in the - channel technology interface. This callback gets called when the - owning channel changes due to a masquerade. Before this was - introduced, if a masquerade happened on a channel being spyed on, - the channel pointer in the datastore became invalid. (closes - issue #12187) (reported by, and lots of testing from atis) (props - to file for the help with ideas) - - * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom - phones to blink when an extension that has been subscribed to - goes on hold. Otherwise, they just stay on like it does when an - extension is in use. (closes issue #11263) Reported by: russell - Patches: notify_hold.rev1.txt uploaded by russell (license 2) - Tested by: russell - - * apps/app_followme.c: Fix a couple uses of sprintf. The second one - could actually cause an overflow of a stack buffer. It's not a - security issue though, it only depends on your configuration. - -2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del - for autocongestion in chan_sip. The scheduler callback will - always return 0. This means that this id is never rescheduled, so - it makes no sense to loop trying to delete the id from the - scheduler queue. If we fail to remove the item from the queue - once, it will fail every single time. (Yes I realize that in this - case, the macro would exit early because the id is set to -1 in - the callback, but it still makes no sense to use that macro in - favor of calling ast_sched_del once and being done with it) This - is the first of potentially several such fixes. - - * include/asterisk/sched.h: Added a large comment before the - AST_SCHED_DEL macro to explain its purpose as well as when it is - appropriate and when it is not appropriate to use it. I also - removed the part of the debug message that mentions that this is - probably a bug because there are some perfectly legitimate places - where ast_sched_del may fail to delete an entry (e.g. when the - scheduler callback manually reschedules with a new id instead of - returning non-zero to tell the scheduler to reschedule with the - same idea). I also raised the debug level of the debug message in - AST_SCHED_DEL since it seems like it could come up quite - frequently since the macro is probably being used in several - places where it shouldn't be. Also removed the redundant line, - file, and function information since that is provided by ast_log. - -2008-03-12 19:57 +0000 [r108135] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c, main/channel.c: (closes issue #12187, - reported by atis, fixed by me after some brainstorming on the - issue with mmichelson) - Update copyright info on app_chanspy. - - Fix a race condition that caused app_chanspy to crash. The issue - was that the chanspy datastore magic that was used to ensure that - spyee channels did not disappear out from under the code did not - completely solve the problem. It was actually possible for - chanspy to acquire a channel reference out of its datastore to a - channel that was in the middle of being destroyed. That was - because datastore destruction in ast_channel_free() was done near - the end. So, this left the code in app_chanspy accessing a - channel that was partially, or completely invalid because it was - in the process of being free'd by another thread. The following - sort of shows the code path where the race occurred: - ============================================================================= - Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) - --------------------------------------||------------------------------------- - ast_channel_free() || - remove channel from channel list || - - lock/unlock the channel to ensure || that no references retrieved - from || the channel list exist. || - --------------------------------------||------------------------------------- - || channel_spy() - destroy some channel data || - Lock chanspy - datastore || - Retrieve reference to channel || - lock channel || - - Unlock chanspy datastore - --------------------------------------||------------------------------------- - - destroy channel datastores || - call chanspy datastore d'tor || - which NULL's out the ds' || - Operate on the channel ... - reference to the channel || || - free the channel || || || - - unlock the channel - --------------------------------------||------------------------------------- - ============================================================================= - -2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: if we receive an INVITE with a - Content-Length that is not a valid number, or is zero, then don't - process the rest of the message body looking for an SDP closes - issue #11475 Reported by: andrebarbosa - -2008-03-12 18:26 +0000 [r108083] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, include/asterisk/audiohook.h, - main/audiohook.c: Add a trigger mode that triggers on both read - and write. The actual function that returns the combined audio - frame though will wait until both sides have fed in audio, or - until one side stops (such as the case when you call Wait). - (closes issue #11945) Reported by: xheliox - -2008-03-12 16:59 +0000 [r108031] Russell Bryant <russell@digium.com> - - * main/channel.c: Destroy the channel lock after the channel - datastores. (inspired by issue #12187) - -2008-03-12 01:52 +0000 [r107877] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql: - Document all of the possible realtime fields - -2008-03-11 23:37 +0000 [r107714-107826] Jason Parker <jparker@digium.com> - - * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql - ODBC voicemail. (closes issue #12186) Reported by: jsmith - Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license - 15) - - * channels/chan_gtalk.c: Copy voicemail dependency logic for - res_adsi to chan_gtalk (for jabber). (closes issue #12014) - Reported by: junky - -2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile.rules, channels/Makefile: get chan_vpb to build properly - in dev mode - -2008-03-11 20:47 +0000 [r107712] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Add a newline on a log - -2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Make sure the visible indication is on the - right channel so when the masquerade happens the proper - indication is enacted. (closes issue #11707) Reported by: iam - - * apps/app_meetme.c: Add an additional check for setting conference - parameter when using the marked user options. It was possible for - it to return to a no listen/no talk state if a masquerade - happened. (closes issue #12136) Reported by: aragon - - * apps/app_exec.c: Fix a minor spelling error. (closes issue - #12183) Reported by: darrylc - -2008-03-11 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.19-rc2 released. - -2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_rpt.c: backport a fix from trunk - - * channels/misdn/isdn_lib.c, codecs/Makefile, - channels/chan_misdn.c: fix various other problems found by gcc - 4.3 - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - apps/app_sms.c: stop checking for mktime() in the configure - script... we don't use it, and the test is buggy under gcc 4.3 - - * configure, main/Makefile, configure.ac, makeopts.in: check for - compiler support for -fno-strict-overflow before using it (tested - with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179) - Reported by: Netview - - * configure, configure.ac: fix small bug in IMAP toolkit testing - - * main/udptl.c, utils/Makefile, main/Makefile, - main/editline/readline.c, pbx/Makefile: fix up various compiler - warnings found with gcc-4.3: - the output of flex includes a - static function called 'input' that is not used, so for the - moment we'll stop having the compiler tell us about unused - variables in the flex source files (a better fix would be to - improve our flex post-processing to remove the unused function) - - main/stdtime/localtime.c makes assumptions about signed integer - overflow, and gcc-4.3's improved optimizer tries to take - advantage of handling potential overflow conditions at compile - time; for now, suppress these optimizations until we can fiure - out if the code needs improvement - main/udptl.c has some - references to uninitialized variables; in one case there was no - bug, but in the other it was certainly possibly for unexpected - behavior to occur - main/editline/readline.c had an unused - variable - -2008-03-11 00:59 +0000 [r107290] Terry Wilson <twilson@digium.com> - - * channels/chan_sip.c: If we fail to alloc a channel, we should - re-lock the pvt structure before returning. - -2008-03-10 21:32 +0000 [r107230] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Use non-global storage for eswitch - -2008-03-10 20:27 +0000 [r107173] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Make sure to reenable echo can after a - "failed" (canceled, etc) three-way call. (closes issue #11335) - Reported by: rebuild - -2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant <russell@digium.com> - - * main/pbx.c: Fix another bug specifically related to asynchronous - call origination. Once the PBX is started on the channel using - ast_pbx_start(), then the ownership of the channel has been - passed on to another thread. We can no longer access it in this - code. If the channel gets hung up very quickly, it is possible - that we could access a channel that has been free'd. (inspired by - BE-386) - - * main/pbx.c: Fix some bugs related to originating calls. If the - code failed to start a PBX on the channel (such as if you set a - call limit based on the system's load average), then there were - cases where a channel that has already been free'd using - ast_hangup() got accessed. This caused weird memory corruption - and crashes to occur. (fixes issue BE-386) (much debugging credit - goes to twilson, final patch written by me) - - * main/channel.c: Resolve a compiler warning. - - * main/channel.c: Fix a race condition where the generator can go - away (closes issue #12175, reported by edantie, patched by me) - -2008-03-10 14:33 +0000 [r107016] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where - unanswered CDRs are dropped to the CDR core, not everything uses - app_dial. (closes issue #11516) Reported by: ys Patches: - branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: - anest, jcapp, dartvader - -2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: don't generate D-Channel "up" and "down" - messages unless the channel state is actually changing; also, - generate the "up" message when an implicit "up" occurs due to - reception of a normal event when we thought the channel was - "down" - -2008-03-07 22:51 +0000 [r106895] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Only start the SLA thread if SLA has actually - been configured. - -2008-03-07 22:14 +0000 [r106842] Jason Parker <jparker@digium.com> - - * main/editline/Makefile.in: Fix hardcoded grep in editline, were - GNU grep is required. (closes issue #12124) Reported by: dmartin - -2008-03-07 19:32 +0000 [r106788] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Ignore source update control frame. (closes issue - #12168) Reported by: plack - -2008-03-07 17:16 +0000 [r106704] Russell Bryant <russell@digium.com> - - * include/asterisk/sched.h: Change a warning message to a debug - message. This is happening quite frequently, and it is not worth - spamming users with these messages unless we are pretty confident - that it should never happen. As it stands today, it _will_ and - _does_ happen and until that gets cleaned up a reasonable amount - on the development side, let's not spam the logs of everyone - else. (closes issue #12154) - -2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Warn the user when a temporary greeting - exists (Closes issue #11409) - - * main/rtp.c: Properly initialize rtp->schedid (Closes issue - #12154) - - * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c, - apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c, - funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, - main/manager.c: Safely use the strncat() function. (closes issue - #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt - uploaded by Corydon76 (license 14) - -2008-03-06 22:10 +0000 [r106437] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: Quell an annoying message that is likely to print - every single time that ast_pbx_outgoing_app is called. The reason - is that __ast_request_and_dial allocates the cdr for the channel, - so it should be expected that the channel will have a cdr on it. - Thanks to joetester on IRC for pointing this out - -2008-03-06 04:40 +0000 [r106328] Tilghman Lesher <tlesher@digium.com> - - * sounds/Makefile: Upgrade to the next release of sounds - -2008-03-05 22:37 +0000 [r106237] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a potential deadlock and a few - different potential crashes. (closes issue #12145, reported by - thiagarcia, patched by me) - -2008-03-05 22:32 +0000 [r106235] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c, - apps/app_dial.c, main/channel.c, channels/chan_phone.c, - main/dial.c, channels/chan_zap.c, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, main/file.c, - channels/chan_alsa.c, apps/app_followme.c, - include/asterisk/frame.h: Add a control frame to indicate the - source of media has changed. Depending on the underlying - technology it may need to change some things. (closes issue - #12148) Reported by: jcomellas - -2008-03-05 21:12 +0000 [r106178] Michiel van Baak <michiel@vanbaak.info> - - * doc/realtime.txt: document var_metric so no bugreports will come - in when it's actually a configuration issue. (issue #12151) - Reported and patched by: caio1982 1.4 patch by me - -2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: when a PRI call must be moved to a different - B channel at the request of the other endpoint, ensure that any - DSP active on the original channel is moved to the new one - (closes issue #11917) Reported by: mavetju Tested by: mavetju - -2008-03-05 15:17 +0000 [r106015] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c, include/asterisk/sched.h: Correctly - initialize retransid in SIP, and ensure that the warning when - failing to delete a schedule entry can actually hit the log. - (closes issue #12140) Reported by: slavon Patches: sch2.patch - uploaded by slavon (license 288) (Patch slightly modified by me) - -2008-03-05 01:52 +0000 [r105932] Russell Bryant <russell@digium.com> - - * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug - that I just noticed in the RTP code. The calculation for setting - the len field in an ast_frame of audio was wrong when G.722 is in - use. The len field represents the number of ms of audio that the - frame contains. It would have set the value to be twice what it - should be. - -2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: In addition to setting the marker bit let's change - our ssrc so they know for sure it is a different source. - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a - new source of audio comes in (such as music on hold) make sure - the marker bit gets set. (closes issue #10355) Reported by: - wdecarne Patches: 10355.diff uploaded by file (license 11) - (closes issue #11491) Reported by: kanderson - -2008-03-04 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.19-rc1 released. - -2008-03-04 04:31 +0000 [r105591] Russell Bryant <russell@digium.com> - - * main/pbx.c: Backport a minor bug fix from trunk that I found - while doing random code cleanup. Properly break out of the loop - when a context isn't found when verify that includes are valid. - -2008-03-03 18:06 +0000 [r105572] Jason Parker <jparker@digium.com> - - * res/snmp/agent.c: Fix type for astNumChannels. (closes issue - #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg - (license 192) - -2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant <russell@digium.com> - - * channels/chan_local.c: In the case of an ast_channel allocation - failure, take the local_pvt out of the pvt list before destroying - it. - - * channels/chan_local.c: Fix a potential memory leak of the - local_pvt struct when ast_channel allocation fails. Also, in - passing, centralize the code necessary to destroy a local_pvt. - - * main/autoservice.c: Update the copyright information for - autoservice. Most of the code in this file now is stuff that I - have written recently ... - - * main/asterisk.c, main/channel.c, include/asterisk.h, - main/autoservice.c: Merge in some changes from - team/russell/autoservice-nochans-1.4 These changes fix up some - dubious code that I came across while auditing what happens in - the autoservice thread when there are no channels currently in - autoservice. 1) Change it so that autoservice thread doesn't keep - looping around calling ast_waitfor_n() on 0 channels twice a - second. Instead, use a thread condition so that the thread - properly goes to sleep and does not wake up until a channel is - put into autoservice. This actually fixes an interesting bug, as - well. If the autoservice thread is already running (almost always - is the case), then when the thread goes from having 0 channels to - have 1 channel to autoservice, that channel would have to wait - for up to 1/2 of a second to have the first frame read from it. - 2) Fix up the code in ast_waitfor_nandfds() for when it gets - called with no channels and no fds to poll() on, such as was the - case with the previous code for the autoservice thread. In this - case, the code would call alloca(0), and pass the result as the - first argument to poll(). In this case, the 2nd argument to - poll() specified that there were no fds, so this invalid pointer - shouldn't actually get dereferenced, but, this code makes it - explicit and ensures the pointers are NULL unless we have valid - data to put there. (related to issue #12116) - -2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp <jcolp@digium.com> - - * main/channel.c: It is possible for no audio to pass between the - current digit and next digit so expand logic that clears - emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: - edgreenberg Patches: v1-11911.patch uploaded by dimas (license - 88) Tested by: tbsky - - * channels/chan_sip.c: Add a comment to describe some logic. - (closes issue #12120) Reported by: flefoll Patches: - chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license - 244) - -2008-02-29 23:34 +0000 [r105409] Russell Bryant <russell@digium.com> - - * main/autoservice.c: Fix a major bug in autoservice. There was a - race condition in the handling of the list of channels in - autoservice. The problem was that it was possible for a channel - to get removed from autoservice and destroyed, while the - autoservice thread was still messing with the channel. This led - to memory corruption, and caused crashes. This explains multiple - backtraces I have seen that have references to autoservice, but - do to the nature of the issue (memory corruption), could cause - crashes in a number of areas. (fixes the crash in BE-386) (closes - issue #11694) (closes issue #11940) The following issues could be - related. If you are the reporter of one of these, please update - to include this fix and try again. (potentially fixes issue - #11189) (potentially fixes issue #12107) (potentially fixes issue - #11573) (potentially fixes issue #12008) (potentially fixes issue - #11189) (potentially fixes issue #11993) (potentially fixes issue - #11791) - -2008-02-29 14:47 +0000 [r105326] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Fix a potential memory leak - -2008-02-29 14:34 +0000 [r105296] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: If the message file does not exist, just - return harmlessly, instead of crashing. (Closes issue #12108) - -2008-02-29 13:48 +0000 [r105261] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Bump up the size of the uniqueid variable. - (closes issue #12107) Reported by: asgaroth - -2008-02-29 13:05 +0000 [r105209] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Automatically create new buddy upon reception - of a presence stanza of type subscribed. (closes issue #12066) - Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by - phsultan (license 73) trunk-12066-1.diff uploaded by phsultan - (license 73) Tested by: ffadaie, phsultan - -2008-02-28 22:23 +0000 [r105116] Russell Bryant <russell@digium.com> - - * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock - tracking code that was discovered by mmichelson. The issue is - that if the lock history array was full, then the functions to - mark a lock as acquired or not would adjust the stats for - whatever lock is at the end of the array, which may not be - itself. So, do a sanity check to make sure that we're updating - lock info for the proper lock. (This explains the bizarre stats - on lock #63 in BE-396, thanks Mark!) - -2008-02-28 21:56 +0000 [r105113] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.debian.asterisk: Update init script for LSB - compat (closes issue #9843) Reported by: ibc Patches: - rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by: - paravoid - -2008-02-28 20:11 +0000 [r105059] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: When using autofill, members who are in use - should be counted towards the number of available members to call - if ringinuse is set to yes. Thanks to jmls who brought this issue - up on IRC - -2008-02-28 19:20 +0000 [r104920-105005] Jason Parker <jparker@digium.com> - - * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into - applications, if we get NULL. This protects against possible - segfaults in applications that may try to use data before - checking length (ast_strdupa'ing it, for example) (closes issue - #12100) Reported by: foxfire Patches: 12100-nullappargs.diff - uploaded by qwell (license 4) - - * channels/chan_skinny.c: According to a video at www.cisco.com, - the 7921G supports 6 line appearances. - -2008-02-28 00:05 +0000 [r104868] Tilghman Lesher <tlesher@digium.com> - - * main/Makefile, build_tools/strip_nonapi: Compatibility fix for - PPC64 (closes issue #12081) Reported by: jcollie Patches: - asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412) - Tested by: jcollie, Corydon76 - -2008-02-27 21:49 +0000 [r104841] Mark Michelson <mmichelson@digium.com> - - * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a - lockable list. This is because in some cases, the ast_dial may - exist in multiple threads due to asynchronous execution of its - application, and I found some cases where race conditions could - exist. 2. Check in ast_dial_join to be sure that the channel - still exists before attempting to lock it, since it could have - gotten hung up but the is_running_app flag on the - ast_dial_channel may not have been cleared yet. (closes issue - #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by - putnopvut (license 60) Tested by: jvandal - -2008-02-27 20:56 +0000 [r104787] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c: Don't loop around infinitely trying to spy on - our own channel, and don't forget to free/detach the datastore - upon hangup of the spy. - -2008-02-27 20:36 +0000 [r104783] Mark Michelson <mmichelson@digium.com> - - * main/file.c: Bump a couple of more buffers up by 2 so that - annoying warnings aren't generated like crazy on every - fileexists_core call. - -2008-02-27 18:15 +0000 [r104704] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: Ensure the session ID can't be 0. - -2008-02-27 17:41 +0000 [r104665] Joshua Colp <jcolp@digium.com> - - * main/file.c: Bump up the buffer by 2. - -2008-02-27 17:33 +0000 [r104625] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get - stuck in an infinite loop without being able to detect that the - calling channel hung up. (closes issue #12076, reported by junky, - patched by me) - -2008-02-27 17:26 +0000 [r104598] Jason Parker <jparker@digium.com> - - * res/res_features.c: Inherit language from the transfering channel - on a blind transfer. (closes issue #11682) Reported by: caio1982 - Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982 - (license 22) Tested by: caio1982, victoryure - -2008-02-27 17:07 +0000 [r104596] Joshua Colp <jcolp@digium.com> - - * main/loader.c: Use the lock (which already existed, it just - wasn't used) on the updaters list to protect the contents instead - of the overall module list lock. (closes issue #12080) Reported - by: ChaseVenters - -2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming <kpfleming@digium.com> - - * main/file.c: fallback to standard English prompts properly when - using new prompt directory layout (closes issue #11831) Reported - by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license - 20) (modified by me to improve code and conform rest of function - to coding guidelines) - -2008-02-27 16:45 +0000 [r104591] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: When we receive a known alarm, make sure - that the unknown alarm flag is not still set to make sure that - when we come back out of alarm, it gets reported in the log and - manager interface (after discussion with tzafrir on the -dev - list) - -2008-02-27 15:52 +0000 [r104536] Joshua Colp <jcolp@digium.com> - - * res/res_smdi.c: Only stop the MWI monitor thread if it was - actually started. (closes issue #12086) Reported by: francesco_r - -2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c: Avoid some recursion in the cleanup code for - the chanspy datastore (closes issue #12076, reported by junky, - patched by me) - - * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as - an alarm. However, Asterisk 1.4 does not know what to do with - these alarms. Only Asterisk 1.6 cares about it. So, if we get an - unknown alarm in chan_zap, don't generate confusing log messages - about it. - -2008-02-26 18:26 +0000 [r104132-104141] Jason Parker <jparker@digium.com> - - * Makefile: Add badshell to .PHONY target (thanks Kevin) - - * Makefile: Since all shells aren't as awesome as bash, we have to - fail if somebody tries to use a literal "~" in DESTDIR. - - * sounds/Makefile: Revert previous abspath change. ...abspath is - new in GNU make 3.81. I feel so...defeated. Must find new fix! - - * sounds/Makefile: Fix a very bizarre issue we were seeing with our - buildbot when using a DESTDIR that wasn't an absolute path (such - as DESTDIR=~/asterisk-1.4). Apparently what was happening, was - that some of the targets were being expanded to the full path, so - $@ ended up being /root/asterisk-1.4/[...]/ rather than - ~/asterisk-1.4/[...]/ It appears that this may be a new "feature" - in GNU make. (*cough* - http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*) - -2008-02-26 00:25 +0000 [r104119] Russell Bryant <russell@digium.com> - - * include/asterisk/smdi.h, apps/app_voicemail.c, - channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample: - Merge changes from team/russell/smdi-1.4 This commit brings in a - significant set of changes to the SMDI support in Asterisk. There - were a number of bugs in the current implementation, most notably - being that it was very likely on busy systems to pop off the - wrong message from the SMDI message queue. So, this set of - changes fixes the issues discovered as well as introducing some - new ways to use the SMDI support which are required to avoid the - bugs with grabbing the wrong message off of the queue. This code - introduces a new interface to SMDI, with two dialplan functions. - First, you get an SMDI message in the dialplan using - SMDI_MSG_RETRIEVE() and then you access details in the message - using the SMDI_MSG() function. A side benefit of this is that it - now supports more than just chan_zap. For example, with this - implementation, you can have some FXO lines being terminated on a - SIP gateway, but the SMDI link in Asterisk. Another issue with - the current implementation is that it is quite common that the - station ID that comes in on the SMDI link is not necessarily the - same as the Asterisk voicemail box. There are now additional - directives in the smdi.conf configuration file which let you map - SMDI station IDs to Asterisk voicemail boxes. Yet another issue - with the current SMDI support was related to MWI reporting over - the SMDI link. The current code could only report a MWI change - when the change was made by someone calling into voicemail. If - the change was made by some other entity (such as with IMAP - storage, or with a web interface of some kind), then the MWI - change would never be sent. The SMDI module can now poll for MWI - changes if configured to do so. This work was inspired by and - primarily done for the University of Pennsylvania. (also related - to issue #9260) - -2008-02-26 00:03 +0000 [r104111] Jason Parker <jparker@digium.com> - - * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris. - (closes issue #12050) Reported by: asgaroth Patches: 12050.patch - uploaded by putnopvut (license 60) - -2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c: This patch fixes some pretty significant - problems with how app_chanspy handles pointers to channels that - are being spied upon. It was very likely that a crash would occur - if the channel being spied upon hung up. This was because the - current ast_channel handling _requires_ that the object is locked - or else it could disappear at any time (except in the owning - channel thread). So, this patch uses some channel datastore magic - on the spied upon channel to be able to detect if and when the - channel goes away. (closes issue #11877) (patch written by me, - but thanks to kpfleming for the idea, and to file for review) - - * main/utils.c: Improve the lock tracking code a bit so that a - bunch of old locks that threads failed to lock don't sit around - in the history. When a lock is first locked, this checks to see - if the last lock in the list was one that was failed to be - locked. If it is, then that was a lock that we're no longer - sitting in a trylock loop trying to lock, so just remove it. - (inspired by issue #11712) - -2008-02-25 21:37 +0000 [r104095] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make it so a users.conf user creates both a - SIP peer and a SIP user. The user will be used for inbound - authentication for the device, and peer will be used for placing - calls to the device. (closes issue #9044) Reported by: queuetue - Patches: sip-gui-friend.diff uploaded by qwell (license 4) - -2008-02-25 21:31 +0000 [r104094] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: If the destination folder is full, don't - delete a message when exiting. (closes issue #12065) Reported by: - selsky Patch by: (myself) - -2008-02-25 20:49 +0000 [r104092] Jason Parker <jparker@digium.com> - - * main/config.c: Allow the use of #include and #exec in situations - where the max include depth was only 1. Specifically, this fixes - using #include and #exec in extconfig.conf. This was basically - caused because the config file itself raises the include level to - 1. I opted not to raise the include limit, because recursion here - could cause very bizarre behavior. Pointed out, and tested by - jmls (closes issue #12064) - -2008-02-25 18:38 +0000 [r104086] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: Ensure that the channel doesn't disappear - in agent_logoff(). If it does, it could cause a crash. (fixes the - crash reported in BE-396) - -2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: If a resubscription comes in for a dialog we - no longer know about tell the remote side that the dialog does - not exist so they subscribe again using a new dialog. (closes - issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff - uploaded by file (license 11) - - * channels/chan_sip.c: Due to recent changes tag will no longer be - NULL if not present so we have to use ast_strlen_zero to see if - it's actually blank. (closes issue #12061) Reported by: flefoll - Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by - flefoll (license 244) - -2008-02-22 22:45 +0000 [r104037] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Backwards debug message. (closes issue - #12052) Reported by: flefoll Patches: - chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license - 244) - -2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c: And as a followup to revision 104026, - completely remove event-related calls from a section of code - where we know there was no event to handle or get. - - * channels/chan_zap.c: Remove an incorrect debug message. It - reported that it had received a specific event and tried to - report which event was received. What actually was happening was - that it was reporting the number of bytes returned from a call to - read(). Thanks to Jared Smith for bringing the issue up on IRC - -2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c: reduce the likelihood that HTTP Manager session - ids will consist of primarily '1' bits - -2008-02-20 22:32 +0000 [r103956] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Clear up confusion when viewing the - QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the - user's perspective, the queue does exist, we shouldn't tell them - we couldn't find the queue. Instead since it is a dead queue, - report a 0 waiting count This issue was brought up on IRC by jmls - -2008-02-20 22:06 +0000 [r103953] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Don't wait for additional digits when - overlap dialing is enabled if the setup message contains the - sending_complete information element. (closes issue #11785) - Reported by: klaus3000 Patches: - sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by - klaus3000 (license 65) - -2008-02-20 21:40 +0000 [r103904] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c: Fix a crash if the channel becomes NULL - while attempting to lock it. (closes issue #12039) Reported by: - danpwi - -2008-02-20 17:53 +0000 [r103845] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/localtime.c: Compat fix for Solaris (closes issue - #12022) Reported by: asgaroth Patches: - 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14) - Tested by: asgaroth - -2008-02-19 20:28 +0000 [r103823] Joshua Colp <jcolp@digium.com> - - * channels/h323/ast_h323.cxx: Send CallerID Name in setup message. - (closes issue #11241) Reported by: tusar Patches: - h323id_as_callerid_name.patch uploaded by tusar (license 344) - -2008-02-19 20:02 +0000 [r103821] Russell Bryant <russell@digium.com> - - * channels/chan_local.c: Account for the fact that the "other" - channel can disappear while the local pvt is not locked. (fixes a - problem introduced in rev 100581) (closes issue #12012) Reported - by: stevedavies Patch by me - -2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Don't look for launchd when cross - compiling. (closes issue #12029) Reported by: ovi - - * channels/chan_sip.c: Fix building of chan_sip. - -2008-02-19 10:27 +0000 [r103806] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Make sure we send error replies correctly by - checking the via header. - -2008-02-18 23:56 +0000 [r103801] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Ensure that emulated DTMFs do not get interrupted - by another begin frame. (closes issue #11740) Reported by: gserra - Patches: v1-11740.patch uploaded by dimas (license 88) (closes - issue #11955) Reported by: tsearle (closes issue #10530) Reported - by: xmarksthespot - -2008-02-18 22:28 +0000 [r103790-103795] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Fix previous commit so that we actually - disable echocanbridged if echocancel is off. - - * channels/chan_zap.c: Correct a message when echocancelwhenbridged - is on, but echocancel is not. Issue #12019 - -2008-02-18 20:52 +0000 [r103786] Mark Michelson <mmichelson@digium.com> - - * main/app.c: There was an invalid assumption when calculating the - duration of a file that the filestream in question was created - properly. Unfortunately this led to a segfault in the situation - where an unknown format was specified in voicemail.conf and a - voicemail was recorded. Now, we first check to be sure that the - stream was written correctly or else assume a zero duration. - (closes issue #12021) Reported by: jakep Tested by: putnopvut - -2008-02-18 17:31 +0000 [r103780] Tilghman Lesher <tlesher@digium.com> - - * main/rtp.c, channels/chan_sip.c: When a SIP channel is being - auto-destroyed, it's possible for it to still be in bridge code. - When that happens, we crash. Delay the RTP destruction until the - bridge is ended. (closes issue #11960) Reported by: norman - Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 - (license 14) Tested by: norman - -2008-02-18 16:37 +0000 [r103770] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c: Fix a linked list corruption that under the - right circumstances could lead to a looped list, meaning it will - traverse forever. (closes issue #11818) Reported by: michael-fig - Patches: 11818.patch uploaded by putnopvut (license 60) Tested - by: michael-fig - -2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Backport fix from issue #9325. (closes issue - #11980) Reported by: rbrunka - - * channels/chan_sip.c: Don't care if the extension given doesn't - exist for subscription based MWI. - -2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race - condition (closes issue #11780) Reported by: guillecabeza - Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license - 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license - 380) - - * main/loader.c: In the case that you try to directly reload a - module has returned AST_MODULE_LOAD_DECLINE, log a message - indicating that the module is not fully initialized and must be - initialized using "module load". - - * main/loader.c: Don't attempt to execute the reload callback for a - module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash - that was reported against chan_console in trunk. (closes issue - #11953, reported by junky, fixed by me) - -2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson <mmichelson@digium.com> - - * doc/imapstorage.txt, configure, configure.ac: Final round of - changes for configure script logic for IMAP Now if a directory is - specified, then we will search that directory for a source - installation of the IMAP toolkit. If none is found, then we will - use that directory as the basis for detecting a package - installation of the IMAP c-client. If that check fails, then - configure will fail. - - * configure, configure.ac: Fix a bit of wrong logic in the - configure script that caused problems when trying to configure - without IMAP. Patch suggestion from phsultan, but I modified it - slightly. (closes issue #12003) Reported by: pj Tested by: - putnopvut - - * doc/imapstorage.txt, configure, configure.ac: I apparently - misunderstood one of the requirements of this configure change. - Now, if a source directory is specified with the --with-imap - option, and a valid source installation is not detected there, - then configure will fail and will not check for a package - installation. - - * doc/imapstorage.txt: Make a small clarification in the - documentation - - * doc/imapstorage.txt: Update documentation regarding configuration - of IMAP - - * apps/app_voicemail.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Change to the - configure logic regarding IMAP. Prior to this commit, if you - wished to configure Asterisk with IMAP support, you would use the - --with-imap configure switch in one of the following two ways: - --with-imap=/some/directory would look in the directory specified - for a UW IMAP source installation --with-imap would assume that - you had imap-2004g installed in .. relative to the Asterisk - source With this set of changes the two above options still work - the same, but there are two new behaviors, too. - --with-imap=system will assume that you have -libc-client.so - where you store your shared objects and will attempt to find - c-client headers in your include path either in the imap or - c-client directory. If either of the two original methods of - specifying the imap option should fail, then the check for - --with-imap =system will be performed in addition. It is only - after this "system" check that failure can happen. - - * apps/app_voicemail.c: Fix build for non-IMAP builds - - * apps/app_voicemail.c: Fix the new message count if delete=yes - when using IMAP storage. (closes issue #11406) Reported by: - jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license - 50) Tested by: jaroth - -2008-02-14 19:51 +0000 [r103683-103684] Jason Parker <jparker@digium.com> - - * funcs/func_cdr.c: swap location for this.. - - * funcs/func_cdr.c: Document the 'l' option to the CDR() function. - (Thanks voipgate for pointing out the option, and Leif for - providing text for it.) Closes issue #11695. - -2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_agent.c: We aren't talking to ourselves; we're - talking to someone else. (closes issue #11771) Reported by: - msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982 - (license 22) Tested by: caio1982, msetim - - * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is - not loaded (which is a symbol dependency) (closes issue #11760) - Reported by: non-poster Patches: 20080114__bug11760.diff.txt - uploaded by Corydon76 (license 14) Tested by: Corydon76, - non-poster, jamesgolovich - -2008-02-12 22:24 +0000 [r103503-103504] Jason Parker <jparker@digium.com> - - * main/asterisk.c: revert accidental change from last commit. oops - - * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition - that was impossible. - -2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Even if no CallerID name or number has been - provided by the remote party still use the configured sip.conf - ones. (closes issue #11977) Reported by: pj - - * apps/app_meetme.c: If entering a conference with the 'w' option - ensure that we can't listen or speak until the marked user - appears. (closes issue #11835) Reported by: alanmcmillan - -2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming <kpfleming@digium.com> - - * configs/zapata.conf.sample: improve 2BCT documentation a bit - (thanks Jared) - -2008-02-09 06:23 +0000 [r103197] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Commit fix for being unable to send - voicemail from VoiceMailMain Reported by: William F Acker (via - the -users mailing list) Patch by: Corydon76 (license 14) - -2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Prevent a potential three-thread deadlock. Also - added a comment block to explicitly state the locking order - necessary inside app_queue. (closes issue #11862) Reported by: - flujan Patches: 11862.patch uploaded by putnopvut (license 60) - Tested by: flujan - - * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl - fails for Zaptel timing device. (closes issue #11891) Reported - by: tzafrir - -2008-02-08 15:08 +0000 [r102968] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Make sure the presence of dbsecret is - factored into user scoring. (closes issue #11952) Reported by: - bbhoss - -2008-02-07 19:53 +0000 [r102858] Jason Parker <jparker@digium.com> - - * res/res_features.c: Specify which digit string was matched in - debug message. (closes issue #11949) Reported by: dimas Patches: - v1-feature-debug.patch uploaded by dimas (license 88) - -2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming <kpfleming@digium.com> - - * configs/zapata.conf.sample: document usage of 'transfer' - configuration option for ISDN PRI switch-side transfers - -2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only consider a T.38-only INVITE compatible - if we have both a joint capability between us and them and if - they provided T.38. - - * main/global_datastores.c: Add missing header file and - ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by: - snuffy - -2008-02-06 15:19 +0000 [r102651] Russell Bryant <russell@digium.com> - - * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so - that it gets inherited by outbound channels. (due to a discussion - between me and a user via email) - -2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming <kpfleming@digium.com> - - * pbx/Makefile, res/Makefile: ensure that all remaining - multi-object modules are built using their proper CFLAGS and - include directory paths - -2008-02-06 00:26 +0000 [r102576] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Move around some defines to unbreak ODBC - storage. (closes issue #11932) Reported by: snuffy - -2008-02-05 20:02 +0000 [r102453] Mark Michelson <mmichelson@digium.com> - - * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes - issue #11919) Reported by: eferro Patches: - mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) - Tested by: eferro - -2008-02-05 19:52 +0000 [r102450] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: If a REGISTER attempt comes in that is a - retransmission of a previous REGISTER do not create a new nonce - value. (issue #BE-381) - -2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming <kpfleming@digium.com> - - * channels/Makefile: ensure that components of chan_misdn.so are - built using any special build options that the configure script - generated (reported by Philipp Kempgen on asterisk-dev) - -2008-02-05 15:09 +0000 [r102378] Joshua Colp <jcolp@digium.com> - - * res/res_clioriginate.c: Perform dialing asynchronously when using - the originate CLI command so the CLI does not appear to block. - (closes issue #11927) Reported by: bbhoss - -2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, utils/muted.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Cross-platform - fix: OS X now deprecates the use of the daemon(3) API. (closes - issue #11908) Reported by: oej Patches: - 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76 - - * funcs/func_strings.c: Missing braces. (closes issue #11912) - Reported by: dimas Patches: sprintf.patch uploaded by dimas - (license 88) - -2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE - (according to RFC 3261) (closes issue #9492) Reported by: - kryptolus Patches: bug9492.txt uploaded by oej (license 306) - Tested by: oej - - * channels/chan_sip.c: Handle ACK and CANCEL in an invite - transaction - even if we get INFO transactions during the actual - call setup. (closes issue #10567) Reported by: jacksch Tested by: - oej Patch by: oej inspired by suggestions from neutrino88 in the - bug tracker - -2008-02-01 23:06 +0000 [r101989] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns - out that even though G.722 is 16 kHz, it is supposed to specified - as 8 kHz in the RTP, and RTP timestamps are supposed to be - calculated based on 8 kHz. (Apparently this is due to a bug in a - spec, but people follow it anyway, because it's the spec ...) - -2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded - voicemail, and fixed several other bugs while I'm in the area. - (closes issue #11615) Reported by: jamessan Patches: - 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76, jamessan - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: Change detection of getifaddrs to use - AST_C_COMPILE_CHECK, backported from trunk (as suggested by - kpfleming) - -2008-02-01 17:41 +0000 [r101822] Jason Parker <jparker@digium.com> - - * apps/app_authenticate.c: Remove a needless (and incorrect) call - to feof() after fgets(). This would have exited the loop early if - you had an authentication file with no newline at the end. - -2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant <russell@digium.com> - - * apps/app_authenticate.c: off by one error - - * apps/app_authenticate.c: Don't overwrite the last character of a - line if it's not a newline. This would happen if the last line in - the file doesn't have a newline. (pointed out by Qwell) - -2008-02-01 15:55 +0000 [r101772] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/acl.c: Compatibility fix for OpenWRT (reported by Brian - Capouch via the mailing list) - -2008-02-01 00:32 +0000 [r101693] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial - strings for the case that no peer or hostname was provided, which - is the one part of the dial string that is absolutely required. - If it's not there, bail out. (closes issue #11897) Reported by - sokhapkin Patch by me - -2008-02-01 00:06 +0000 [r101649] Mark Michelson <mmichelson@digium.com> - - * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle - periods of no channel activity" (closes issue #9256) Reported by: - cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt - uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, - rjain - -2008-01-31 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.18 released. - -2008-01-31 23:10 +0000 [r101601] Russell Bryant <russell@digium.com> - - * main/translate.c, main/file.c: Fix a couple of places where - ast_frfree() was not called on a frame that came from a - translator. This showed itself by g729 decoders not getting - released. Since the flag inside the translator frame never got - unset by freeing the frame to indicate it was no longer in use, - the translators never got destroyed, and thus the g729 licenses - were not released. (closes issue #11892) Reported by: xrg - Patches: 11892.diff uploaded by russell (license 2) Tested by: - xrg, russell - -2008-01-31 21:00 +0000 [r101531] Mark Michelson <mmichelson@digium.com> - - * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the - beginning of an absolute pathname. 2. If ast_monitor_change_fname - is called and the new filename is the same as the old, then exit - early and don't set the filename_changed field in the monitor - structure. Setting it in this case was causing ast_monitor_stop - to erroneously delete them. (closes issue #11741) Reported by: - garlew Tested by: putnopvut - -2008-01-31 19:52 +0000 [r101482] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes - for struct in_addr funkiness. Issue #11885, patch by snuffy. - -2008-01-31 19:30 +0000 [r101480] Steve Murphy <murf@digium.com> - - * main/pbx.c: closes issue #11845; that's the one where there's a - 1004 byte cdr leak with every AMI Redirect to a zap channel - -2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: Add more missing locking of the agents - list ... - - * channels/chan_agent.c: Move the locking from find_agent() into - the agent dialplan function handler to ensure that the agent - doesn't disappear while we're looking at it. - - * channels/chan_agent.c: Add missing locking to the find_agent() - function. - -2008-01-30 15:41 +0000 [r101222] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Fix an issue where if a frame of higher - sample size preceeded a frame of lower sample size and - ast_slinfactory_read was called with a sample size of the - combined values or higher a crash would happen. (closes issue - #11878) Reported by: stuarth - -2008-01-30 15:34 +0000 [r101219] Jason Parker <jparker@digium.com> - - * configs/extensions.conf.sample: Change default config to use - descending channel order of groups, rather than ascending. Fixes - a potential source of confusion in glare-type situations. Issue - 11875, reported by JimVanM. - -2008-01-30 15:23 +0000 [r101216] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix a logic error with regards to autofill. - Prior to this change, it was possible for a caller to go out of - turn if autofill were enabled and callers ahead in the queue were - attempting to call a member. This change fixes this. - -2008-01-30 11:20 +0000 [r101152] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Stop musiconhold on attended transfer. - (closes issue #11872) Reported by: gareth Patches: - svn-101018.patch uploaded by gareth (license 208) - -2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard <dhubbard@digium.com> - - * build_tools/make_version: updated build_tools to handle the - autotag directory structure changes; changes related to BE-353. - Patch by The Russell and reviewed by The Me. - -2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Remove a memory leak from updating realtime - queues - - * apps/app_queue.c: Fixing an erroneous return value returned when - attempting to pause or unpause a queue member fails. Fixes - BE-366, thanks to John Bigelow for writing the patch. - -2008-01-29 17:57 +0000 [r100934] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c: Don't forget to record the channel so we - know whether it is bridged or not later. (closes issue #11811) - Reported by: slavon - -2008-01-29 17:43 +0000 [r100932] Russell Bryant <russell@digium.com> - - * main/Makefile: Fix the last couple of issues related to building - from a path that contains spaces. (closes issue #11834) - -2008-01-29 17:41 +0000 [r100930] Jason Parker <jparker@digium.com> - - * channels/misdn_config.c: Initialize an array to 0s if config - option not specified. (closes issue #11860) Patches: - misdn_get_config.v1.diff uploaded by IgorG (license 20) - -2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant <russell@digium.com> - - * Makefile: Use GNU make magic instead of shell magic to escape - spaces in the working directory. (related to issue #11834) - - * Makefile: Fix building Asterisk when the working path has spaces - in it. (closes issue #11834) Reported by: spendergrass Patched - by: me - -2008-01-29 16:10 +0000 [r100835] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Allow zap groups above 30 to work properly. - (closes issue #11590) Reported by: tbsky - -2008-01-29 10:36 +0000 [r100793] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed potential segfault in misdn show - channels CLI command - -2008-01-29 08:26 +0000 [r100740] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: (closes issue #11736) Reported by: MVF - Patches: bug11736-2.diff uploaded by oej (license 306) Tested by: - oej, MVF, revolution (russellb: This was the showstopper for the - release.) - -2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly - (went to 't' instead of 'T') - -2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson@digium.com> - - * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of - chan_vpb. It is alive and well. - -2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put - greeting files into the database like we do with the others. - Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded - by dimas (license 88) - -2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: For some reason, the use of this strdupa() - is leading to memory corruption on freebsd sparc64. This trivial - workaround fixes it. (closes issue #10300, closes issue #11857, - reported by mattias04 and Home-of-the-Brave) - - * res/res_features.c: Fix a crash in ast_masq_park_call() (issue - #11342) Reported by: DEA Patches: res_features-park.txt uploaded - by DEA (license 3) - -2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Correct a comment which made little/no - sense. - -2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell@digium.com> - - * main/channel.c, channels/chan_local.c, - include/asterisk/channel.h: Make some deadlock related fixes. - These bugs were discovered and reported internally at Digium by - Steve Pitts. - Fix up chan_local to ensure that the channel lock - is held before the local pvt lock. - Don't hold the channel lock - when executing the timing function, as it can cause a deadlock - when using chan_local. This actually changes the code back to be - how it was before the change for issue #10765. But, I added some - other locking that I think will prevent the problem reported - there, as well. - -2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher@digium.com> - - * main/rtp.c, channels/chan_mgcp.c, main/cdr.c, - channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c, - channels/chan_h323.c, include/asterisk/sched.h, main/file.c, - pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from - the scheduler, ignoring the return value could possibly cause - memory to be accessed after it is freed, which causes all sorts - of random memory corruption. Instead, if a deletion fails, wait a - bit and try again (noting that another thread could change our - taskid value). (closes issue #11386) Reported by: flujan Patches: - 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76, flujan, stuarth` - -2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson@digium.com> - - * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is - now preferred that users of Voicetronix products use chan_zap in - combination with their zaptel drivers. - -2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: This would have never been true, since we're - passing (sizeof(req.data) - 1) as the len to recvfrom(). - -2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/app.h: make these macros not assume that the - only other field in the structure is 'argc'... this is true when - someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable - to define your own structure as long as it has the right fields - -2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Update main Asterisk copyright info to 2008 - -2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker@digium.com> - - * main/acl.c: Fix compilation on Solaris. (closes issue #11832) - Patches: bug-11832.diff uploaded by snuffy (license 35) - -2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Second attempt. Don't change invitestate - when receiving 18x messages in CANCEL state. (issue #11736) - Reported by: MVF Patch by oej. - - * channels/chan_sip.c: Make sure we don't cancel destruction on - calls in CANCEL state, even if we get 183 while waiting for - answer on our CANCEL. (issue #11736) Reported by: MVF Patches: - bug11736.txt uploaded by oej (license 306) Tested by: MVF - -2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson@digium.com> - - * apps/app_externalivr.c: Fixing a typo. - -2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the - beginning of a list of channels to potentially spy on. However, - if there were no matching channels, it would beep at you over and - over, which is pretty annoying. Now, it will only beep once in - the case that there are no channels to spy on, but it will still - beep again once it reaches the beginning of the channel list - again. (closes issue #11738, patched by me) - -2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: These flag tests were illogical. They were - testing sip_peer flags on a sip_pvt. Thanks to Russell for - helping to get this odd problem figured out. - -2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: When we reset the password via an external - command, we should also reset the password stored in the - in-memory list, too (otherwise it doesn't really take effect). - (closes issue #11809) Reported by: davetroy Patches: - fix_externpass.diff uploaded by davetroy (license 384) - - * res/res_odbc.c: Oops, should have checked for a NULL obj, here, - too - - * main/acl.c: Just confirmed that all current platforms need this - header file - -2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Thanks to Russell's education I realize that - BUFSIZ has changed since I learned the C language over 20 years - ago... Resetting chan_sip to the size of BUFSIZ that I expected - in my old head to avoid to heavy memory allocations on some - systems. - -2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher@digium.com> - - * main/acl.c: Fix the defines for OS X (and Solaris, too) - -2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej@edvina.net> - - * channels/chan_local.c, res/res_features.c, channels/chan_agent.c, - apps/app_followme.c: Add more dependencies on chan_local and add - a note to the description of chan_local so that people don't - disable it in menuselect just to clean up. - - * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial - still runs without chan_local, but will be missing forwarding - functionality. - -2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher@digium.com> - - * main/acl.c: Ensure that we can get an address even when we don't - have a default route. (closes issue #9225) Reported by: junky - Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 - (license 14) Tested by: oej, loloski, sergee - -2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Cleaning up some documentation that led to - confusion in a bug report - -2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c: Fixing an issue wherein monitoring local - channels was not possible. During a channel masquerade, the - monitors on the two channels involved are swapped. In 99% of the - cases this results in the desired effect. However, if monitoring - a local channel, this caused the monitor which was on the local - channel to get moved onto a channel which is immediately hung up - after the masquerade has completed. By swapping the monitors - prior to the masquerade, we avoid the problem by tricking the - masquerade into placing the monitor back onto the channel where - we want it. During the investigation of the issue, the channel's - monitor was the only thing that was swapped in such a manner - which did not make sense to have done. All other variable - swapping made sense. - -2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c, configs/res_odbc.conf.sample, - include/asterisk/res_odbc.h: Permit the user to specify number of - seconds that a connection may remain idle, which fixes a crash on - reconnect with the MyODBC driver. (closes issue #11798) Reported - by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt - uploaded by Corydon76 (license 14) Tested by: mvanbaak - -2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Bump the buffer size for Via headers up to - 512. There are some exceptionally large Via headers out there. - (closes issue #11783) Reported by: ofirroval - -2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell@digium.com> - - * main/slinfactory.c: Fix a couple of memory leaks with frame - handling. Specifically, ast_frame_free() needed to be called on - the frame that came from the translator to signed linear. - -2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp@digium.com> - - * include/asterisk/channel.h: Remove the __ in front of the unused - variable. This causes some compilers to freak out. - -2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell@digium.com> - - * include/asterisk/translate.h, main/frame.c: Revert adding the - packed attribute, as it really doesn't make sense why that would - do any good. Fix the real bug, which is to do the check to see if - the frame came from a translator at the beginning of - ast_frame_free(), instead of at the end. This ensures that it - always gets checked, even if none of the parts of the frame are - malloc'd, and also ensures that we aren't looking at free'd - memory in the case that it is a malloc'd frame. (closes issue - #11792, reported by explidous, patched by me) - - * include/asterisk/translate.h: Since we're relying on the offset - between the frame and the beginning of the translator pvt struct, - set the packed attribute to make sure we get to the right place. - (potential fix for issue #11792) - -2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson@digium.com> - - * res/res_features.c: This should at least temporarily fix a - problem where the 't' Dial option is incorrectly passed to the - transferee when built-in attended transfers are used. There is - still a problem with 'T', but better to fix some problems than no - problems while we work on it. (closes issue #7904) Reported by: - k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee - (license 138) Tested by: sergee, otherwiseguy - -2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand@digium.com> - - * configs/cdr.conf.sample: doh! revert a revert of a revert - (changed by mistake in 99010) - - * main/manager.c, configs/cdr.conf.sample: missed that one while - reverting - - * main/manager.c: reverting 99001 - We need the Max-Age for - extending the life of cookie mansession_id - -2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell@digium.com> - - * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: - Have IAX2 optimize the codec translation path just like chan_sip - does it. If the caller's codec is in our codec list, move it to - the top to avoid transcoding. (closes issue #10500) Reported by: - stevedavies Patches: iax-prefer-current-codec.patch uploaded by - stevedavies (license 184) iax-prefer-current-codec.1.4.patch - uploaded by stevedavies (license 184) Tested by: stevedavies, pj, - sheldonh - -2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c: we should only send the Set-Cookie header to the - browser on the first response after creating a manager session, - not on every response (doing so causes the browser to clear any - local cookies it may have associated with the session) - -2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker@digium.com> - - * configs/zapata.conf.sample: Add a clarification about the - immediate= option of zapata.conf Issue 11784, patch by klaus3000. - -2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell@digium.com> - - * .cleancount, include/asterisk/channel.h: Add an unused pointer to - the ast_channel struct. This makes the ast_channel structure - retain the same size as it had in previous 1.4 releases. Also, - all of the offsets for members in the structure are still the - same (except for the two pointers that got replaced for the new - spy/whisper architecture.) - -2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp@digium.com> - - * .cleancount: Bump up cleancount due to previous commit that - changed the channel structure. - - * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c, - main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h - (added), main/Makefile, include/asterisk/chanspy.h (removed), - include/asterisk/channel.h, main/audiohook.c (added): Replace - current spy architecture with backport of audiohooks. This should - take care of current known spy issues. - - * channels/chan_iax2.c: Add missing NULLs at end of two - ast_load_realtimes. (closes issue #11769) Reported by: tequ - Patches: chaniax.patch uploaded by dimas (license 88) - -2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c: Fix a deadlock in chan_local in - local_hangup. There was contention because the local_pvt was held - and it was attempting to lock a channel, which is the incorrect - locking order. (closes issue #11730) Reported by: UDI-Doug - Patches: 11730.patch uploaded by putnopvut (license 60) Tested - by: UDI-Doug - -2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp@digium.com> - - * main/dial.c: Introduce a lock into the dialing API that protects - it when destroying the structure. (closes issue #11687) Reported - by: callguy Patches: 11687.diff uploaded by file (license 11) - - * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes - issue #11777) Reported by: tootai - - * channels/chan_sip.c: Don't drop the old record route information - when dealing with packets related to a reinvite. (closes issue - #11545) Reported by: kebl0155 Patches: reinvite-patch.txt - uploaded by kebl0155 (license 356) - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, codecs/codec_speex.c, - configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later - versions use a separate library for some things so we need to use - it if present in codec_speex. (closes issue #11693) Reported by: - yzg - -2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Change a buffer in check_auth() to be a - thread local dynamically allocated buffer, instead of a massive - buffer on the stack. This fixes a crash reported by Qwell due to - running out of stack space when building with LOW_MEMORY defined. - On a very related note, the usage of BUFSIZ in various places in - chan_sip is arbitrary and careless. BUFSIZ is a system specific - define. On my machine, it is 8192, but by definition (according - to google) could be as small as 256. So, this buffer in - check_auth was 16 kB. We don't even support SIP messages larger - than 4 kB! Further usage of this define should be avoided, unless - it is used in the proper context. - - * main/rtp.c, include/asterisk/translate.h, main/frame.c, - main/translate.c, main/abstract_jb.c, channels/chan_iax2.c, - codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for - some memory access errors pointed out by the valgrind2.txt output - on issue #11698. The issue here is that it is possible for an - instance of a translator to get destroyed while the frame - allocated as a part of the translator is still being processed. - Specifically, this is possible anywhere between a call to - ast_read() and ast_frame_free(), which is _a lot_ of places in - the code. The reason this happens is that the channel might get - masqueraded during this time. During a masquerade, existing - translation paths get destroyed. So, this patch fixes the issue - in an API and ABI compatible way. (This one is for you, - paravoid!) It changes an int in ast_frame to be used as flag - bits. The 1 bit is still used to indicate that the frame contains - timing information. Also, a second flag has been added to - indicate that the frame came from a translator. When a frame with - this flag gets released and has this flag, a function is called - in translate.c to let it know that this frame is doing being - processed. At this point, the flag gets cleared. Also, if the - translator was requested to be destroyed while its internal frame - still had this flag set, its destruction has been deffered until - it finds out that the frame is no longer being processed. - Admittedly, this feels like a hack. But, it does fix the issue, - and I was not able to think of a better solution ... - -2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Based on the boundary found move over the - correct amount. (closes issue #11750) Reported by: tasker - - * channels/chan_sip.c: Accept "; boundary=" not just ";boundary=" - in the multipart mixed content type. (closes issue #11750) - Reported by: tasker - -2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Adding in appropriate unlocks for the locks - I added. Thanks to joetester on IRC for pointing this out. - -2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell@digium.com> - - * main/translate.c: Revert a change that introduces an unacceptable - performance hit and is causing memory leaks ... (from rev 97973) - -2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixing another compilation error. I'm a bit off - today :( - - * apps/app_queue.c: Oops. Last commit had compilation error. - - * apps/app_queue.c: Adding explicit defaults for missing options to - init_queue. This is necessary because if a user either removes or - comments one of these options and reloads their queues, the - option will not reset to its default, instead maintaining the - value from prior to the reload. Thanks to John Bigelow for - pointing this error out to me. - -2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c: Add a connection timeout attribute, as that was - what was intended with the login timeout, but ODBC divides it up - into 2 different timeouts. (Closes issue #11745) - -2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems. - (closes issue #11646) Reported by: caio1982 Patches: - dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) - dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested - by: caio1982, mvanbaak - -2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand@digium.com> - - * main/http.c: Comment explaining how to force browser to always - read some html files from server. - -2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: If the incoming RTP stream changes codec force the - bridge to break if the other side does not support it. (closes - issue #11729) Reported by: tsearle Patches: - new_codec_patch_udiff.patch uploaded by tsearle (license 373) - - * res/res_agi.c: If the channel is hungup during RECORD FILE send a - result code of -1 to be uniform with everything else. (closes - issue #11743) Reported by: davevg Patches: res_agi.diff uploaded - by davevg (license 209) - -2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: Properly report the hangup cause as no answer - when someone does not answer (closes issue #10574, reported by - boch, patched by moy) - -2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher@digium.com> - - * codecs/gsm/Makefile: Add another exception (which doesn't work) - for -march optimization flag. Reported by: thomasmebes Patch by: - tilghman (Closes issue #11563) - -2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell@digium.com> - - * doc/security.txt, main/asterisk.c, configure, - include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, - makeopts.in: Backport the ability to set the ToS bits on Linux - when not running as root. Normally, we would not backport - features into 1.4, but, I was convinced by the justification - supplied by the supplier of this patch. He pointed out that this - patch removes a requirement for running as root, thus reducing - the potential impacts of security issues. (closes issue #11742) - Reported by: paravoid Patches: libcap.diff uploaded by paravoid - (license 200) - -2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp@digium.com> - - * apps/app_followme.c: Ensure the return value of ast_bridge_call - is passed back up as the application return value. This is needed - for transfers to function so the PBX core knows to continue - execution. (closes issue #10327) Reported by: kkiely - -2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Back out changes from revision 97077, since - it wasn't perfect - -2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell@digium.com> - - * main/frame.c: Fix samples vs. length calculations for g722 - - * main/translate.c: Simplify this code with a suggestion from Luigi - on the asterisk-dev list. Instead of using is16kHz(), implement a - format_rate() function. - - * main/translate.c: Fix various timing calculations that made - assumptions that the audio being processed was at a sample rate - of 8 kHz. - -2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c, main/translate.c: 1) When we get a - translated frame out, clone it, because if the translator pvt is - freed before we use the frame, bad things happen. 2) Getting a - failure from ast_sched_delete means that the schedule ID is - currently running. Don't just ignore it. (Closes issue #11698) - -2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Let us leave a voicemail for ourself if we - have logged into VoiceMailMain and chosen to leave a message. - (closes issue #11735, reported and patched by jamessan) - -2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf@digium.com> - - * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the - same fixes for ael.flex as was done in 97849 for ast_expr2.fl; - overrode the normally generate yyfree func with our own version - that checks the pointer for non-null before passing to free(). - Also takes care of a little problem with 2.5.33 and the use of - the __STDC_VERSION__ macro. - - * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a - fix for 2 things: a problem Terry was having in OSX with null - pointers, which was my fault, as I probably forgot to run the sed - script last time I made mods. So, I moved the fix into the flex - input itself. Then, I found when I used flex 2.5.33, that it was - using __STDC_VERSION__, and that's not real good; so I added back - in a DIFFERENT sed script to fix that little mess. Tested - everything, a couple different ways. Hope I did no harm, at the - least. - -2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker@digium.com> - - * include/asterisk/frame.h: Fix a comment that is no longer true. - -2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell@digium.com> - - * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample, - pbx/kdeconsole_main.cc (removed): Remove other remnants of - pbx_kdeconsole - - * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in, - configure, include/asterisk/autoconfig.h.in, configure.ac, - makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't - worked in ages, and nobody has complained. (closes issue #11706, - reported by caio1982) - -2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp@digium.com> - - * funcs/func_groupcount.c: Don't try to copy the category from the - group if no category exists. (closes issue #11724) Reported by: - IgorG Patches: group_count.v1.patch uploaded by IgorG (license - 20) - -2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell@digium.com> - - * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose - messages - - * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't - actually load on my system still (related to issue #11706) - -2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker@digium.com> - - * main/cli.c: Correctly display a message if a command could not be - found. Also fix a comment which may have led to this happening. - Issue 11718, reported by kshumard. - - * main/cli.c: Fix some locking and return value funkiness. We - really shouldn't be unlocking this lock inside of a function, - unless we locked it there too. - -2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some - smaller functions this time - -2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell@digium.com> - - * res/res_features.c: Fix saying the parking space number to the - caller doing the parking ... - -2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: report the same message whether Zaptel does - not have transcoder support loaded or no transcoders were found - -2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc - function. As underlined in issue #10437 by Josh, we need to - prevent a possible memory leak. We only set the name part of the - caller id, the number part is not relevant when dealing with - JIDs. Closes issue #11549. - -2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Don't do conferencing totally in Zaptel if - Monitor is running on the channel. (closes issue #11709) Reported - by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy - (license 371) - -2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: pass the right variable to get an error - string... oops - - * channels/chan_zap.c: add error number output to ioctl failure - messages to help with debugging - -2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher@digium.com> - - * main/cli.c, main/editline/readline.c: Allow filename completion - on zero-length modules, remove a memory leak, remove a file - descriptor leak, and make filename completion thread-safe. - Patched and tested by tilghman. (Closes issue #11681) - -2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: use the \retval doxygen command properly - - * apps/app_queue.c: Part 1 of N of adding doxygen comments to - app_queue. I picked some of the most common functions used (which - also happen to be some the biggest/ugliest functions too) to - document first. I'm pretty new to doxygen so criticism is - welcome. - - * apps/app_queue.c: Some coding guidelines-related cleanup - -2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp@digium.com> - - * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp. - (closes issue #11443) Reported by: eferro Patches: - dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license - 337) - -2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher@digium.com> - - * main/autoservice.c, main/utils.c: Increase constants to where - we're less likely to hit them while debugging. (Closes issue - #11694) - -2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Making some changes designed to not allow - for a corrupted mailstream for a vm_state. 1. Add locking to the - vm_state retrieval functions so that no linked list corruption - occurs. 2. Make sure to always grab the persistent vm_state when - mailstream access is necessary. 3. Correct an incorrect return - value in the init_mailstream function. (closes issue #11304, - reported by dwhite) - -2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp@digium.com> - - * funcs/func_groupcount.c: If no group has been provided to the - GROUP_COUNT dialplan function then use the first one specific to - the channel. (closes issue #11077) Reported by: m4him - - * apps/app_queue.c: Make app_queue calls work with directed pickup. - (closes issue #11700) Reported by: jbauer - -2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes, - found in issue #11386, but not completely closing that issue. - -2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 96931 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | - 2 lines Change misery.digium.com to pbx.digium.com ........ - - * res/res_smdi.c: Don't crash if something happens when setting up - an SMDI interface and it gets destroyed before the SMDI port - handling thread gets created. - -2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Indentation fix, makes the code easier to read - - * res/res_jabber.c: Compute the base64 value over the - [authzid]\0authcid\0password string, thus excluding the trailing - NULL byte. This change has already been committed to trunk, see - #11644. - -2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell@digium.com> - - * main/devicestate.c: Don't pass an empty string as the device - name. - -2008-01-04 23:03 +0000 [r96575] Tilghman Lesher <tlesher@digium.com> - - * main/devicestate.c: Fix the problem of notification of a device - state change to a device with a '-' in the name. Could probably - do with a better fix in trunk, but this bug has been open way too - long without a better solution. Reported by: stevedavies Patch - by: tilghman (Closes issue #9668) - -2008-01-04 22:55 +0000 [r96573] Jason Parker <jparker@digium.com> - - * res/res_features.c: Properly continue in the dialplan if using - PARKINGEXTEN and the slot is full. Issue 11237, patch by me. - -2008-01-04 19:27 +0000 [r96525] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: If you change the bindaddr in sip.conf to a - non-bound address and reload, sip goes kablooie. Reported and - patched by: one47 (Closes issue #11535) - -2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Make use of the temporary channel pointer - while the pvt is unlocked. (closes issue #11675) Reported by: - flefoll Patches: chan_zap.c.patch-store-owner-before-unlock - uploaded by flefoll (license 244) - - * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has - been destroyed before we get to this point (closes issue #11672, - reported by snuffy, patched by me) - -2008-01-03 21:37 +0000 [r96318] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c: Missed initialization caused crash. - Reported and fixed by: tiziano (Closes issue #11671) - -2008-01-03 12:12 +0000 [r96198-96199] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: make sure frame is completely clean, - before we send it to asterisk as DTMF. If we don't make it clean, - it happens that one way audio occurs.. - - * channels/chan_misdn.c: when overlapdial was used and no number - was dialed, the call was dropped, now we just jump into the s - extension, which makes a lot more sense. - -2008-01-02 23:46 +0000 [r96102] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: We need to reset the membername to NULL on each - iteration of this loop, otherwise the result is that multiple - members can have the same name, since the variable was not reset - on each iteration of the loop. - -2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: Convert locks of the contexts list in - pbx_config to the appropriate rdlock or wrlock - - * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts - list. - - * apps/app_macro.c: app_macro only needs a rdlock on the contexts - list. - -2008-01-02 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.17 released. - -2008-01-02 20:24 +0000 [r95946] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Allocate a SIP refer structure when - performing a transfer using BYE with Also so that the transfer - information is properly stored. (AST-2008-028) (closes issue - #11637) Reported by: greyvoip - -2008-01-02 17:51 +0000 [r95890] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: A change to improve the accuracy of queue - logging in the case where a member does not answer during the - specified timeout period. Prior to this change, there was a small - chance that the member name recorded in this case would be blank. - Also prior to this change, if using the ringall strategy, if no - one answered the call during the specified timeout, the member - name listed in the queue log would randomly be one of the members - that was rung. (closes issue #11498, reported and tested by - hloubser, patched by me) - -2007-12-31 23:43 +0000 [r95577] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: Avoiding a potentially bad locking situation. - ast_merge_contexts_and_delete writelocks the conlock, then calls - ast_hint_extension, which attempts to readlock the same lock. - Recursion with read-write locks is dangerous, so the inner lock - needs to be removed. I did this by copying the "guts" of - ast_hint_extension into ast_merge_contexts_and_delete (sans the - extra lock). (this change is inspired by the locking problems - seen in issue #11080, but I have no idea if this is the - problematic area experienced by the reporters of that issue) - -2007-12-31 20:27 +0000 [r95470] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_env.c: Allow the default "0" to be returned if the - STAT fails (Closes issue #11659) - -2007-12-28 18:24 +0000 [r95191] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove duplicate increment of the header - count in the add_header() function. (closes issue #11648) - Reported by: makoto Patch provided by sergee, committed patch by - me, inspired by comments from putnopvut - -2007-12-28 00:16 +0000 [r95095] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: I found a bug while browsing the queue code and - managed to reproduce it in a small setup. If a queue uses the - ringall strategy, it was possible through unfortunate coincidence - for a single member at a given penalty level to make app_queue - think that all members at that penalty level were unavailable and - cause the members at the next penalty level to be rung. With this - patch, we will only move to the next penalty level if ALL the - members at a given penalty level are unreachable. - -2007-12-27 21:40 +0000 [r95024] Russell Bryant <russell@digium.com> - - * main/channel.c: Don't report a syntax error when an empty string - is passed to ast_get_group. Just return 0. (closes issue #11540) - Reported by: tzafrir Patches: group_empty.diff uploaded by - tzafrir (license 46) -- slightly changed by me - -2007-12-27 20:09 +0000 [r94977] Mark Michelson <mmichelson@digium.com> - - * main/io.c: Fixing a typo in a comment. - -2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Include types.h in chan_h323 as without it - it can not be compiled on some operating systems like FreeBSD to - name one. (closes issue #11585) Reported by: sobomax Patches: - chan_h323.c.diff uploaded by sobomax (license 359) - - * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is - set or not on the dialog. It is possible for it to be a pointer - to NULL. (closes issue #11557) Reported by: FuriousGeorge - -2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant <russell@digium.com> - - * main/pbx.c: Now that the contexts lock is a read/write lock, it - should not be locked here in ast_hint_state_changed(). This makes - it get locked recursively which now causes a deadlock. (closes - issue #11080, thanks to callguy for the access to a deadlocked - machine) - - * include/asterisk/translate.h, main/translate.c: Use the constant - that I really meant to use here ... - - * main/translate.c: Change ast_translator_best_choice() to only pay - attention to audio formats. This fixes a problem where Asterisk - claims that a translation path can not be found for channels - involving video. (closes issue #11638) Reported by: cwhuang - Tested by: cwhuang Patch suggested by cwhuang, with some - additional changes by me. - -2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c: make this comment explain the situation in an - even more explicit fashion - -2007-12-26 20:43 +0000 [r94808] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: Workaround for what is probably a glibc bug (but - we'll see this crop up again and again, if we don't add the - workaround). Reported by: rolek Patch by: tilghman (Closes issue - #11601, closes issue #11426) - -2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant <russell@digium.com> - - * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag - was already set before starting autoservice, remember it and - ensure that the channel has the same setting when autoservice - gets stopped. (pointed out by d1mas, patched up by me) - - * main/autoservice.c: When a channel is in autoservice, mark a flag - on the channel that says that we only care about the END of a - digit. That way, no magic digit emulation stuff will happen when - all we're doing is queueing up END frames. - - * res/res_features.c: Don't try to send a parked call back to - itself. (closes issue #11622, reported by djrodman, patched by - me) - - * main/autoservice.c: Don't store DTMF BEGIN frames while a channel - is in autoservice. It's just going to make ast_read() do a lot of - extra work when the channel comes back out of autoservice. - (closes issue #11628, patched by me) - - * Makefile: List include/asterisk/version.h as a .PHONY target - because we want the commands listed for this target to be - executed regardless of whether the file exists or not. This fixes - having the version not up to date when running from svn. (closes - issue #11619, reported by plack, fixed by me) - -2007-12-25 02:27 +0000 [r94769] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: file says... build on the builders. - -2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c: Race: we need to wait to queue a NewChannel event - until after the channel is inserted into the channel list. The - reason is because some manager users immediately queue requests - from the channel when they see that event and are confused when - Asterisk reports no such channel. (Closes issue #11632) - - * channels/chan_sip.c: More deadlock avoidance code (this time - between sip_monitor and sip_hangup) Reported by: apsaras Patch - by: tilghman (Closes issue #11413) - - * channels/chan_sip.c: Another bit of bad logic in realtime_peer - Reported by: dimas Patch by: dimas (Closes issue #11631) - -2007-12-23 01:21 +0000 [r94660] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Argh... I suppose third time's the charm. - -2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Bunch of coding guidelines cleanup - - * apps/app_voicemail.c: Better quota support for using IMAP storage - voicemail (closes issue #11415, reported by jaroth) (closes issue - #11152, reported by selsky) Patch provided by jaroth - - * apps/app_voicemail.c: The mail_copy c-client function does not - expect a full imap mailbox string, just the name of the mailbox. - (closes issue #11419, reported and patched by jaroth, with - additional patchwork from me) - - * main/dial.c: Since we are freeing list elements within a list - traversal, we need to use the safe traversal and remove the item - from the list before freeing it. (closes issue 11612, reported by - dtyoo) - -2007-12-21 16:37 +0000 [r94466] Russell Bryant <russell@digium.com> - - * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to - a read/write lock to resolve a deadlock. This has a nice side - benefit of improving performance. :) (closes issue #11609) - (closes issue #11080) - -2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Removing a debug message I accidentally just - committed - - * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying - dates and times. Also some coding guidelines cleanup. (closes - issue #11599, reported and patched by caio1982, coding guidelines - cleanup by me) - -2007-12-21 15:07 +0000 [r94418] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: Fix for restart-as-user problem reported via the - -dev list - -2007-12-20 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.16.2 released. - -2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) | - 5 lines Fix another potential seg fault ... (closes issue #11606) - Reported by: dimas ........ - - * channels/chan_zap.c: Fix a deadlock in d-channel handling in - chan_zap. This deadlock was introduced by the fix to ensure that - channels are properly locked when handling channel variables. - There were sections of this code where the channel pvt was locked - before the channel lock, when in fact it _must_ be the other way - around. (closes issue #11582) Reported by: bugi - -2007-12-19 23:02 +0000 [r94122] Mark Michelson <mmichelson@digium.com> - - * res/res_monitor.c: Sox versions 13.0.0 and newer do not have - "soxmix" and instead use sox -m. res_monitor needs to use this if - the user does not have soxmix. (closes issue #11589, reported by - amessina, patch inspired by amessina but with a flourish from me) - -2007-12-19 22:48 +0000 [r94077] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: Check - for the existence of the soxmix application on the target - platform and have the result available in autoconfig.h. (part of - issue #11589) - -2007-12-19 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.16.1 released. - -2007-12-19 17:29 +0000 [r93955] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is - NULL. - -2007-12-19 17:04 +0000 [r93949] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't - defined (Closes issue #11602) - -2007-12-18 22:42 +0000 [r93764] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: FreeBSD also does not have byte swap - functions. Issue 11586, patch by sobomax. - -2007-12-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.16 released. - -2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions - 93667 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) - | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........ - -2007-12-18 17:02 +0000 [r93625] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: Rework deadlock avoidance used in ast_write, - since it meant that agent channels which were being monitored had - one audio file recorded and one empty audio file saved. (closes - issue #11529, reported by atis patched by me) - -2007-12-17 22:56 +0000 [r93381-93420] Jason Parker <jparker@digium.com> - - * main/translate.c: What was I thinking when I wrote this - masterpiece? -1 + 1 = 0.. who woulda thunk it?. - -2007-12-17 22:28 +0000 [r93377] Joshua Colp <jcolp@digium.com> - - * main/utils.c: Do not try to access information about a lock when - printing out a trylock attempt. It is possible for the lock that - it references to no longer be valid. This would have caused - segfaults or deadlocks. (issue #BE-263) (closes issue #11080) - Reported by: callguy (closes issue #11100) Reported by: callguy - -2007-12-17 21:12 +0000 [r93336] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/time.h: Today is tomorrow's yesterday, and - yesterday's tomorrow is today, and tomorrow's tomorrow is the day - after tomorrow, so who cares if you recycle anyway? If this - confuses you, that's nothing compared to what this fixes. ;-) - -2007-12-17 19:53 +0000 [r93291] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: We need to create the directory for a - voicemail user even if they are using IMAP storage since - greetings are stored in the filesystem. (closes issue #11388, - reported by spditner, patch by me inspired by a patch by - spditner) - -2007-12-17 18:05 +0000 [r93250] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: If a call is received with a called number - IE containing nothing go to the 's' extension. (closes issue - #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt - uploaded by Corydon76 (license 14) - -2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming <kpfleming@digium.com> - - * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile, - res/Makefile, channels/Makefile, formats/Makefile: fix some - copy-and-paste leftovers - -2007-12-17 07:15 +0000 [r93182] Olle Johansson <oej@edvina.net> - - * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c, - apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add - dependencies on res_monitor and res_features. I wonder if - Asterisk can run at all without res_features. My guess is that - there's propably a lot of more modules and the core that depends - on it. Reported by: caio1982 (closes issue #11574) - -2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming <kpfleming@digium.com> - - * formats, Makefile, codecs/Makefile, funcs, apps/Makefile, - configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in, - formats/Makefile, pbx, res, channels, funcs/Makefile, codecs, - include/asterisk/autoconfig.h.in, build_tools/make_version, apps, - configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps - (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In - http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, - rizzo brought up some issues related to the way that the metadata - required for menuselect and the rest of the build system is - extracted from the source files. Since I had a few hours to kill - on an airplane today, I decided to improve this situation... so - now the system caches the extracted metadata and uses it to build - the menuselect 'tree' as much as it can. The result of this is - that when a single source file is changed, only the metadata for - that file needs to be extracted again, and the rest is used from - the cache files. I also reduced the number of forked processes - required to do the metadata extraction; it was actually possible - to do most of what we needed in the Makefiles themselves without - using any shell scripts at all! On my laptop, these changes - resulted in an 80% decrease in the time required for the - 'menuselect.makeopts' automatic check to occur after editing a - single source file. While doing this work I also cleaned up a few - minor things in the Makefiles, adding a check for 'awk' to the - configure script and changed all remaining places we use 'grep' - or 'awk' to use the ones found by the configure script, and - changed the 'prep_tarball' script to build the menuselect - metadata so that tarballs of Asterisk will include it and won't - require the user to wait while it is extracted after unpacking. - -2007-12-14 17:36 +0000 [r93000] Russell Bryant <russell@digium.com> - - * main/config.c: There are a lot of existing systems that #include - non-existent files. So, to make the transition to treating this - as an error a bit less painless, just issue a huge error message - for now. Then, later, we can reinstate the code that treats it as - a failure. (Thanks to philippel for the feedback) - -2007-12-14 15:16 +0000 [r92937] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Up the length of the format on the SIP - channel since it can now be rather long. (closes issue #11552) - Reported by: francesco_r - -2007-12-14 15:05 +0000 [r92934] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS - state setting and overlap_task thread starting. - -2007-12-14 15:01 +0000 [r92933] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Change help documentation to match actual behavior - (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman - (Closes issue #11548) - -2007-12-14 01:24 +0000 [r92875] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS, - don't spam the CLI with messages about possible deadlocks. - Instead just print the intended single message every five - seconds. (closes issue 11537, reported and patched by dimas) - -2007-12-13 21:28 +0000 [r92815] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c: Properly initialize polarity statuses, so - that they are detected properly. Reported by: julianjm Patch by: - julianjm (Closes issue #10238) - -2007-12-13 20:13 +0000 [r92809] Jason Parker <jparker@digium.com> - - * main/pbx.c: Make application help text a little more clear about - the use of extensions in a filename. - -2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Prevent another potential fd leak - - * apps/app_voicemail.c: Prevent a possible fd leak. - -2007-12-13 00:11 +0000 [r92696] Jason Parker <jparker@digium.com> - - * main/config.c, channels/chan_sip.c, channels/chan_h323.c, - channels/chan_iax2.c: If a typo is found in a config file, we - previous continued on with what was already loaded. We do not - want to do this (see bug below for details). This makes it so - that if a [ is found without a ], the entire config will fail, - and nothing in it will be loaded. Isue #10690. - -2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: emit a warning message when we drop a G.729B - CNG frame destined for the transcoder - -2007-12-12 21:15 +0000 [r92617] Jason Parker <jparker@digium.com> - - * apps/app_meetme.c: Don't increment user count until after name - has been recorded (if enabled). Issue 11048, tested by pep. - -2007-12-12 19:40 +0000 [r92556] Russell Bryant <russell@digium.com> - - * res/res_features.c: resolve compiler warning - -2007-12-12 17:46 +0000 [r92510] Mark Michelson <mmichelson@digium.com> - - * res/res_features.c: Correctly detect where a dynamic feature was - activated. Before this patch, the channel which initiated the - bridge was always assumed to have been the one which activated - the dynamic feature. This patch corrects this. (closes issue - #11529, reported and patched by nic_bellamy) - -2007-12-12 16:52 +0000 [r92463] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: Test - directly for the API that fixed AST-2007-026, to ensure that - older versions of PostgreSQL are no longer acceptable. (Closes - issue #11526) - -2007-12-12 16:08 +0000 [r92443] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Removing an unused variable. - -2007-12-11 19:51 +0000 [r92363] Joshua Colp <jcolp@digium.com> - - * main/global_datastores.c: Fix potential memory leak with the - dialed interfaces list if another memory allocation fails. - (closes issue #11507) Reported by: eliel Patches: - global_datastores.c.patch uploaded by eliel (license 64) - -2007-12-11 17:42 +0000 [r92323] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixing autofill to be more accurate. - Specifically, if calls ahead of the current caller were ringing - members (but not yet bridged) there could be available members - and waiting callers who would not get matched up. The member - availability checker was correctly determining the number of - available members in this scenario, but the queue itself did not - parallelly reflect this status on the pending calls. This commit - corrects the issue. (closes issue #11459, reported by - equissoftware, patched by me) - -2007-12-10 16:36 +0000 [r92204] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Add G729A as another possible payload name for G729. - Some devices use this instead of G729, which is perfectly normal - since the payload number itself is defined and can't be used by - anything else so the name doesn't matter that much. (closes issue - #11483) Reported by: revolution Patches: rtp.diff uploaded by - revolution (license 346) - -2007-12-10 16:29 +0000 [r92202] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: If there are no members in a queue, then the - loop where the datastore for detecting duplicate dialed numbers - will be skipped, meaning the datastore isn't created. This means - that when we try to free it, there's a crash. This stops that - crash from occurring. (closes issue #11499, reported by slavon, - patched by eliel) - -2007-12-10 16:13 +0000 [r92200] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: It is possible for nativeformats to contain - more then one codec, so print out multiple ones. (closes issue - #11366) Reported by: ovi - -2007-12-10 14:04 +0000 [r92158] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Avoid reinvite race situations with two - Asterisks trying to reinvite each other in 1.4 and trunk. This - patch implements support for the 491 error code that Asterisk 1.4 - generates on situations where we get an incoming INVITE and - already has one in progress. Thanks to mavetju for reporting and - to Raj Jain for an excellent explanation of the problem. Patch by - myself. Tested with 8 Asterisk servers connected to each other in - a training network. Closes issue #10481 - -2007-12-07 23:29 +0000 [r91890] Jason Parker <jparker@digium.com> - - * main/dsp.c: We need to make sure we free the input frame if we - return a different frame in ast_dsp_process. Issue 11273, pointed - out by dimas, with a patch by eliel. - -2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: even though Asterisk explicitly requests that - endpoints using G.729 do *not* use Annex B (silence detection and - comfort noise generation) some do anyway; the transcoder card - interface does not currently work properly with CNG frames, so - trim off the CNG before sending the data - -2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant <russell@digium.com> - - * main/utils.c: Make the lock protecting each thread's list of - locks it currently holds recursive. I think that this will fix - the situation where some people have said that "core show locks" - locks up the CLI. (related to issue #11080) - - * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS - code. The ast_mutex_init() function had the mutex attribute - object marked as static. This means that multiple threads - initializing locks at the same time could step on each other and - end up with improperly initialized locks. (found when tracking - down locking issues related to issue #11080) - - * include/asterisk/lock.h: I love fixing lock related errors in the - lock debugging code. That's about as ironic as it gets in - Asterisk programming land. Anyway, I spotted this bug while - trying to track down why systems are locking up and acting weird - in issue #11080. The mutex attribute object was marked as static - in this function when it should not have been. - - * apps/app_dial.c: * Add channel locking around datastore - operations that expect the channel to be locked. * Document why - we don't record Local channels in the dialed interfaces list. * - Remove the dialed variable as it isn't needed. * Restructure some - code for clarity and coding guidelines stuff - - * apps/app_queue.c: * Add channel locking around datastore - operations that expect the channel to be locked. * Document why - we don't record Local channels in the dialed interfaces list. * - Handle memory allocation failure. * Remove the dialed variable, - as it wasn't actually needed. * Tweak some formatting to conform - to coding guidelines. - - * main/autoservice.c: * Add a bit more of a verbose comment as to - why a hangup frame needs to be queued up if autoservice gets a - NULL return from ast_read(). * Make the process of queueing the - hangup frame more efficient by putting the frame where it is - going to end up and avoiding some locking and extra memory - allocations and freeing. - -2007-12-07 15:39 +0000 [r91737] Mark Michelson <mmichelson@digium.com> - - * main/autoservice.c: Hangups that happen during autoservice were - not processed appropriately. This is because a hangup actually - causes a NULL frame to be received, not a hangup frame. Queueing - a hangup if we receive a NULL frame during autoservice corrects - this problem (closes issue #11467, reported by jmls, patched by - me) - -2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant <russell@digium.com> - - * apps/app_dial.c: Don't unlock the dialed_interfaces list until - we're done messing with the iterator. - - * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels - from Queue() and Dial() again. There was a slight flaw in the - code to prevent call forwards from looping that caused this - problem. (related to issue #11486) - - * apps/app_queue.c: Fix in an issue in the call forwarding handling - code that was causing crashes on every call into a queue. I'm not - entirely sure about the logic in this part of the code, so I want - to look at it some more tomorrow. However, this makes it safe and - keeps it from crashing. (closes issue #11486, reported by adamg, - patched by me) - -2007-12-07 00:52 +0000 [r91637] Tilghman Lesher <tlesher@digium.com> - - * main/rtp.c: At the end of a call, when we're reporting, RTCP may - already be partially torn down, so check for NULL dereference - Reported by: blitzrage Patch by: tilghman (Closes issue #11450) - -2007-12-06 20:25 +0000 [r91541] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: IMAP storage did not honor the maxmsg - setting in voicemail.conf, and it also had the possibility of - crashing if a user had more than 256 messages in their voicemail. - This patch kills two birds with one stone by adding maxmsg - support and also setting a hard limit on the number of messages - at 255 so that the crashes cannot happen. (closes issue #11101, - reported by Skavin, patched by me) - -2007-12-06 19:11 +0000 [r91501] Russell Bryant <russell@digium.com> - - * main/loader.c, include/asterisk/module.h: Add a new module flag - to indicate that a build sum is present. Modules built against - older Asterisk 1.4 headers will now load properly with just a - warning indicating that they are old and may cause problems. - (patch by paravoid) - -2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp <jcolp@digium.com> - - * main/udptl.c: Fix various in the udptl implementation. It could - return empty modem frames, have an incorrect sequence number on - packets, and display the wrong sequence number in the debug - messages. (closes issue #11228) Reported by: Cache Patches: - udptl-4.patch uploaded by dimas (license 88) - - * channels/chan_sip.c: Add support for accepting and sending T.38 - in the initial INVITE. (closes issue #9402) Reported by: thdei - -2007-12-06 12:54 +0000 [r91366] Olle Johansson <oej@edvina.net> - - * main/loader.c, include/asterisk/logger.h, main/logger.c: Make - sure logger is reloaded at general reload in the cli. (Discovered - during Asterisk training in Portugal) - -2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Reverting extra stuff I didn't mean to - commit - - * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial() - did not properly handle the case where only a label was provided. - This was due to the fact that the answering channel did not have - an extension set, so ast_parseable_goto would fail. This fix - eliminates the call to ast_parseable_goto on the answering - channel since it is a wasteful call. The answering channel and - the calling channel are both directed to the same extension and - context, just different priorities, so we can just copy the - values from the calling channel to the answering channel and - increment the answering channel's priority. (closes issue #11382, - reported by jon, patch by me with correction by jon) - -2007-12-05 21:38 +0000 [r91237] Tilghman Lesher <tlesher@digium.com> - - * sounds/Makefile: Upgrade to the latest version of extra sounds - -2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant <russell@digium.com> - - * main/threadstorage.c: Make the lock in the threadstorage - debugging code untracked to avoid a deadlock on thread - destruction. (closes issue #11207) Reported by: ys Patches: - threadstorage.c.diff uploaded by ys (license 281) Also fixes an - open bug report: (closes issue #11446) - - * main/utils.c: When DEBUG_THREADS is enabled, we only have the - details about who is holding a lock that we are waiting on for a - mutex, not rwlocks. This should fix the problem where people have - reported "core show locks" crashing sometimes. - - * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were - reported as happening on Mac systems. It turns out that the - problem was the Mac version of the ast_atomic_fetchadd_int() - function. The Mac atomic add function returns the _new_ value, - while this function is supposed to return the old value. So, the - crashes happened on unreferencing objects. If the reference count - was decreased to 1, ao2_ref() thought that it had been decreased - to zero, and called the destructor. However, there was still an - outstanding reference around. (closes issue #11176) (closes issue - #11289) - - * include/asterisk/file.h, configure, - include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/compiler.h: Modify file.h to maintain API - compatibility with earlier versions. If a recent compiler is - being used, then a warning will show up for any modules still - using the old name "private" instead of "_private". (patch - suggested by paravoid) - - * main/pbx.c: Make some changes to some additions I made recently - for doing channel autoservice when looking up extensions. This - code was added to handle the case where a dialplan switch was in - use that could block for a long time. However, the way that I - added it, it did this for all extension lookups. However, lookups - in the in-memory tree of extensions should _not_ take long enough - to matter. So, move the autoservice stuff to be only around - executing a switch. - -2007-12-04 17:28 +0000 [r90876] Jason Parker <jparker@digium.com> - - * main/channel.c: If we fail to create a channel after allocating a - timing fd, we need to make sure to close it. Issue 11454, patch - by eliel. - -2007-12-04 05:29 +0000 [r90798] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Fix build issue on the build cluster. - -2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/compat.h: Solaris requires the inclusion of - sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: - snuffy,tilghman (Closes issue #11430) - - * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a - NULL deref could result Reported by: xrg Patch by: tilghman - (Closes issue #11387) - -2007-12-03 23:12 +0000 [r90735] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, main/channel.c, main/global_datastores.c - (added), channels/chan_local.c, main/Makefile, - include/asterisk/channel.h, include/asterisk/global_datastores.h - (added), apps/app_queue.c: A big one... This is the merge of the - forward-loop branch. The main change here is that call-forwards - can no longer loop. This is accomplished by creating a datastore - on the calling channel which has a linked list of all devices - dialed. If a forward happens, then the local channel which is - created inherits the datastore. If, through this progression of - forwards and datastore inheritance, a device is attempted to be - dialed a second time, it will simply be skipped and a warning - message will be printed to the CLI. After the dialing has been - completed, the datastore is detached from the channel and - destroyed. This change also introduces some side effects to the - code which I shall enumerate here: 1. Datastore inheritance has - been backported from trunk into 1.4 2. A large chunk of code has - been removed from app_dial. This chunk is the section of code - which handles the call forward case after the channel has been - requested but before it has been called. This was removed because - call-forwarding still works fine without it, it makes the code - less error-prone should it need changing, and it made this set of - changes much less painful to just have the forwarding handled in - one place in each module. 3. Two new files, global_datastores.h - and .c have been added. These are necessary since the datastore - which is attached to the channel may be created and attached in - either app_dial or app_queue, so they need a common place to find - the datastore info. This approach was taken in case similar - datastores are needed in the future, there will be a common place - to add them. - -2007-12-03 22:06 +0000 [r90696] Jason Parker <jparker@digium.com> - - * apps/app_meetme.c: Make sure we always close the conference fd if - we have an open one. Issue 11383, reported by markmhy, patch by - eliel. - -2007-12-03 20:59 +0000 [r90639] Mark Michelson <mmichelson@digium.com> - - * channels/chan_mgcp.c: Changing some bad logic when calculating - the interdigit timeout. (closes issue #11402, reported and - patched by eferro) - -2007-12-03 20:51 +0000 [r90607] Jason Parker <jparker@digium.com> - - * res/res_features.c: Fix crash in ParkAndAnnounce application. - Issue #11436, reported by lytledd, patch by eliel. - -2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Do not create a smoother for G723.1 frames, they need - to be left alone to their native 20/24 byte size. - - * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve - the indication currently playing on a channel when a masquerade - operation happens. (issue #BE-88) - -2007-12-03 18:20 +0000 [r90546] Jason Parker <jparker@digium.com> - - * channels/chan_iax2.c: Only log debug messages if debug is - enabled. Closes issue #11416, patch by casper. - -2007-12-02 18:18 +0000 [r90470] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: The other day when I went through making - changes as a result of the ao2_link() change, I added some code - to set pointers to NULL after they were unreferenced. This - pointed out that in this place, the object was unreferenced - before the code was done using it. So, move the unref down a - little bit. (crash reported by jmls on IRC) - -2007-12-02 09:34 +0000 [r90432] Tilghman Lesher <tlesher@digium.com> - - * main/autoservice.c: Clarify the return value on autoservice. - Specifically, if you started autoservice and autoservice was - already on, it would erroneously return an error. Reported by: - adiemus Patch by: dimas (Closes issue #11433) - -2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant <russell@digium.com> - - * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h, - apps/app_queue.c, channels/chan_iax2.c: Change the behavior of - ao2_link(). Previously, in inherited a reference. Now, it - automatically increases the reference count to reflect the - reference that is now held by the container. This was done to be - more consistent with ao2_unlink(), which automatically releases - the reference held by the container. It also makes it so it is no - longer possible for a pointer to be invalid after ao2_link() - returns. - - * include/asterisk/astobj2.h: Add some notes on the behavior of - ao2_unlink() after a discussion with Tilghman - -2007-11-30 14:43 +0000 [r90269] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix locking issues under one legged replaces - scenarios. (closes issue #11420) Reported by: irroot Patches: - chan_sip_oneleg.patch uploaded by irroot (license 52) - -2007-11-30 00:16 +0000 [r90231] Mark Michelson <mmichelson@digium.com> - - * channels/chan_mgcp.c: Clear the DTMF buffer if the call times - out. (closes issue #11418, reported and patched by eferro) - -2007-11-29 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.15 released. - -2007-11-29 19:48 +0000 [r90166] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure - we use thread-safe escaping (Fixes AST-2007-026) - -2007-11-29 19:38 +0000 [r90163] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: This patch handles the case where a queue - member with a negative penalty is added via the manager. If a - negative value is submitted for a member penalty, we set it to 0. - (closes issue #11411, reported and patched by Laureano) - -2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c: Properly escape input buffers (Fixes - AST-2007-025) - - * formats/format_g726.c, include/asterisk/file.h, - formats/format_wav.c, formats/format_pcm.c, - formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c, - formats/format_h264.c, formats/format_wav_gsm.c: Use of "private" - as a field name in a header file messes with C++ projects - Reported by: chewbacca Patch by: casper (Closes issue #11401) - - * sounds/Makefile: Upgrade the core sounds release version - -2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant <russell@digium.com> - - * funcs/func_callerid.c: fix some formatting i accidentally changed - - * funcs/func_callerid.c, main/channel.c, - include/asterisk/channel.h: This set of changes is to make some - callerID handling thread-safe. The ast_set_callerid() function - needed to lock the channel. Also, the handlers for the CALLERID() - dialplan function needed to lock the channel when reading or - writing callerid values directly on the channel structure. - - * include/asterisk/file.h, main/file.c: Merge a change from - team/russell/chan_refcount ... This makes ast_stopstream() - thread-safe. - -2007-11-28 22:59 +0000 [r90101] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405) - Reported by: eliel Patches: load_realtime.patch uploaded by eliel - (license 64) - -2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming <kpfleming@digium.com> - - * configs/users.conf.sample, main/manager.c: it is impossible to - set permissions for manager accounts created by users.conf - (reported internally, patched by me) - -2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: Removing some seemingly pointless code. This sets a - channel variable for every priority executed in the dialplan if - you have debug set to anything non-zero. This seems pointless due - to the fact that these channel variables are not referenced - anywhere else in the code and their names are esoteric enough - that they would not be practical to reference in the dialplan. - Plus the fact that this behavior isn't documented anywhere means - that the change is not likely to cause any disruption. If - anything, this may actually cause a slight performance increase - if running with debug on. The motivating influence for this code - change is the eventwhencalled option for queues. If set to vars, - all channel variables will be output to the manager. These - unnecessary channel variables make the output a lot more - difficult to deal with. - - * apps/app_voicemail.c: Recording greetings when using IMAP storage - was causing zero-length files to be stored. Since greetings are - not retrieved from IMAP anyway, it is pointless to attempt - storing them there. (closes issue #11359, reported by spditner, - patched by me) - -2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant <russell@digium.com> - - * main/pbx.c, include/asterisk/pbx.h: - update documentation for - some of the goto functions to note that they handle locking the - channel as needed - update ast_explicit_goto() to lock the - channel as needed - - * main/autoservice.c: Don't do frame processing if ast_read() - returned NULL. - - * apps/app_queue.c: Instead of depending on the return value of - ast_true(), explicitly set the eventwhencalled variable to 1. - - * main/pbx.c: Don't start/stop autoservice in - pbx_extension_helper() unless a channel exists - -2007-11-27 23:10 +0000 [r89837] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Two changes with regards to the - 'eventwhencalled' option of queues.conf 1) Due to some signed vs. - unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' - did exactly the same thing. Thus the sign change of the ast_true - call. 2) The vars2manager function overwrote a \n for every - channel variable it parsed, resulting in bizarre output for the - channel variables. This patch remedies this. (related to issue - #11385, however I'm not sure if this will actually be enough to - close it) - -2007-11-27 21:45 +0000 [r89790] Russell Bryant <russell@digium.com> - - * main/autoservice.c, main/pbx.c: Merge changes from - team/russell/autoservice_1.4 This set of changes fixes an issue - that was reported to me on IRC yesterday. The user, d1mas, was - using chan_zap for incoming calls and was having DTMF recognition - issues in some situations. Specifically, he noticed that the - problem occurred when using DISA or WaitExten. He also noticed - that when using Read, the problem did not occur. His system also - used DUNDi for dialplan lookups. So, he theorized that if the - DUNDi lookups blocked for some period of time, that audio from - the zap channel could get lost. If the audio got lost, then it - wouldn't be run through the DTMF detector, and digits could get - lost. He was correct, and the following set of changes fixes the - problem. However, the changes go a little bit further than what - was necessary to fix this exact problem. 1) I updated - pbx_extension_helper() to autoservice the associated channel to - handle cases where extension lookups may take a long time. This - would normally be a dialplan switch that does some lookup over - the network, such as the DUNDi or IAX2 switches. This ensures - that even while a DUNDi lookup is blocking, the channel will be - continuously serviced. 2) I made a change to the autoservice - code. This is actually something that has bothered me for a long - time. When a channel is in autoservice, _all_ frames get thrown - away. However, some frames really shouldn't be thrown away. The - most notable examples are signalling (CONTROL) frames, and DTMF. - So, this patch queues up important frames while a channel is in - autoservice. When autoservice is stopped on the channel, the - queued up frames get stuck back on the channel so that they can - get processed instead of thrown away. 3) I made another change to - the autoservice code to handle the case where autoservice is - started on channels recursively. Previously, you could call - ast_autoservice_start() multiple times on a channel, and it would - stop the first time ast_autoservice_stop() gets called. Now, it - will ensure that autoservice doesn't actually stop until the - final call to ast_autoservice_stop(). - -2007-11-27 20:22 +0000 [r89727] Mark Michelson <mmichelson@digium.com> - - * res/res_config_pgsql.c: Changing some calls from free() to - ast_free() since they were allocated with ast_calloc(). (closes - issue #11390, reported and patched by Laureano) - -2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming <kpfleming@digium.com> - - * main/app.c: on second thought... revert all the other changes - i've made in app options parsing leaving only one: if an empty - argument is supplied for an option, set that argument pointer to - point to an empty string rather than NULL, so that the - application can do normal checks on it without worrying about it - being NULL - - * main/app.c: generate a warning when an application option that - requires an argument is ignored due to lack of an argument - -2007-11-27 16:12 +0000 [r89634] Russell Bryant <russell@digium.com> - - * configs/voicemail.conf.sample: Add a note to the sample voicemail - config noting that when using IMAP storage, only the first format - specified will be attached to the message. - -2007-11-27 15:38 +0000 [r89631] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_env.c: Default result of STAT should be "0" not "". - Reported via the -users mailing list, fixed by me. - -2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson <oej@edvina.net> - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we - get a codec offer using a well-known payload type, but using it - for another codec that we don't know, Asterisk did not remove - that codec from the list. With this patch, we remove the codec - from audio and video rtp objects and deny it ever existed. Thanks - to lasse for testing. (closes issue #11376) Reported by: lasse - Patches: bug11376.txt uploaded by oej (license 306) Tested by: - lasse - - * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue - #11304) Reported by: pj - -2007-11-27 06:24 +0000 [r89622] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample, - include/asterisk/cdr.h: closes issue #11379; OK, this is an - attempt to make both sides happy. To the cdr.conf file, I added - the option 'unanswered', which defaults to 'no'. In this mode, - you will see a cdr for a call, whether it was answered or not. - The disposition will be NO ANSWER or ANSWERED, as appropriate. - The src is as you'd expect, the destination channel will be one - of the channels from the Dial() call, usually the last in the - list if more than one chan was specified. With unanswered set to - 'yes', you will still see this cdr entry in both cases. But in - the case where the dial timed out, you will also see a cdr for - each line attempted, marked NO ANSWER, with no destination - channel name. The new option defaults to 'no', so you don't see - the pesky extra cdr's by default, and you will not see the - irritating 'not posted' messages. - -2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson <mmichelson@digium.com> - - * apps/app_playback.c: After issuing a "say load new", if a caller - hangs up during the middle of playback of a number, app_playback - will continue to try to play the remaining files. With this - change, no more files will be played back upon hangup. (closes - issue #11345, reported and patched by IgorG) - - * apps/app_playback.c: After issuing a "say load new" tons of - warning messages are printed out to the CLI every time do_say in - app_playback is called. Removing these warnings - -2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp <jcolp@digium.com> - - * main/dial.c: Fix issues with async dialing with an application - executing. The application has to be terminated and control - returned to the thread before hanging things up. (issue #BE-252) - - * res/res_features.c: Add module counting removal for error - conditions. (closes issue #11333) Reported by: Laureano Patches: - res_features_v2.c.patch uploaded by Laureano (license 265) - -2007-11-26 17:41 +0000 [r89594] Russell Bryant <russell@digium.com> - - * main/pbx.c: Add channel locking to a function that needed to be - doing it. This is just a little something I noticed while working - on a completely unrelated issue. - -2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_config.c: Use ast_free to free memory, or else we shall - implode if MALLOC_DEBUG is enabled. (closes issue #11347) - Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys - (license 281) - - * apps/app_mixmonitor.c: Close the audio file before sending it to - the post processing application. (closes issue #11357) Reported - by: reformed Patches: mixmonitor.patch uploaded by reformed - (license 330) - -2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming <kpfleming@digium.com> - - * main/app.c: when parsing application options that take arguments, - don't indicate that the option was supplied unless a - non-zero-length argument was found for it - -2007-11-26 15:48 +0000 [r89580] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Revert vmu->email back to an empty string - if it was empty when imap_store_file was called. This prevents - sending a duplicate e-mail. (closes issue #11204, reported by - spditner, patched by me) - -2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp <jcolp@digium.com> - - * main/channel.c: If channel allocation fails because the alert - pipe could not be created also free the scheduler context. - (closes issue #11355) Reported by: eliel Patches: - main.channel.c.patch uploaded by eliel (license 64) - - * apps/app_meetme.c: When unloading app_meetme destroy any auto - created contexts created by SLA. (closes issue #11367) Reported - by: eliel - -2007-11-25 17:17 +0000 [r89559] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c, configs/res_odbc.conf.sample, - include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously - attempted to use the ESCAPE clause to set the escape delimiter to - a backslash. Unfortunately, this does not universally work on all - databases, since on databases which natively use the backslash as - a delimiter, the backslash itself needs to be delimited, but on - other databases that have no delimiter, backslashing the - backslash causes an error. So the only solution that I can come - up with is to create an option in res_odbc that explicitly - specifies whether or not backslash is a native delimiter. If it - is, we use it natively; if not, we use the ESCAPE clause to make - it one. Reported by: elguero Patch by: tilghman (Closes issue - #11364) - -2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher <tlesher@digium.com> - - * res/res_adsi.c: Free some frames that would otherwise leak on - error. Reported by: Laureano Patch by: Laureano,tilghman (Closes - issue #11351) - - * apps/app_voicemail.c, main/app.c: Currently, zero-length - voicemail messages cause a hangup in VoicemailMain. This change - fixes the problem, with a multi-faceted approach. First, we do - our best to avoid these messages from being created in the first - place, and second, if that fails, we detect when the voicemail - message is zero-length and avoid exiting at that point. Reported - by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083) - - * main/manager.c: Up until this point, the XML output of the - manager has been technically invalid, due to the repetition of - certain parameters in a single event. This caused various issues - for XML parsers, some of which refused to parse at all, given the - invalidity of the rendered XML. So this commit fixes the XML - output, ensuring that each entity parameter has a unique name, - thus ensuring valid XML. Reported by: msetim Patch by: tilghman - (Closes issue #10220) - - * res/res_config_odbc.c: Use ESCAPE clause for the first parameter, - not just 2nd-Nth parameters. Reported by: apsaras Patch by: - tilghman (Closes issue #11353) - -2007-11-22 17:29 +0000 [r89527] Russell Bryant <russell@digium.com> - - * configs/agents.conf.sample: mvanbaak pointed out a spelling error - in this sample configuration file. While I was at it, I went - ahead and tweaked it a little bit more. - -2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix a small error I made in my previous commit - - * apps/app_queue.c: Changing an inaccurate debug message to be less - inaccurate. Under the circumstances, this message would always - report that there were 0 members available, even though that may - not be true. - -2007-11-21 18:59 +0000 [r89491] Terry Wilson <twilson@digium.com> - - * res/res_features.c: If a channel gets masqueraded in the middle - of a park, don't play the announcement to the masqueraded - channel, and dial back to the original channel on timeout. - -2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/module.h: re-doxygen some comments - - * main/loader.c, include/asterisk/module.h, - build_tools/make_buildopts_h: bring back compile-option checking - when loading modules, only this time use a string-based storage - and comparison mechanism because it is easier to support on other - platforms - -2007-11-20 17:50 +0000 [r89457] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: According to comments in main/pbx.c, it is essential - that if we are going to lock the conlock as well as the hints - lock, it must be locked in that respective order. In order to - prevent a potential deadlock, we need to lock the conlock prior - to locking the hints lock in ast_hint_state_changed (see the call - stack example on issue #11323 for how this can happen). (closes - issue #11323, reported by eelcob, suggestion for patch by eelcob, - patch by me) - -2007-11-20 15:22 +0000 [r89450] Steve Murphy <murf@digium.com> - - * doc/queues-with-callback-members.txt: closes issue #11324; break - statements missing in switch cases. - -2007-11-20 13:40 +0000 [r89445] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: added RR patch from iroot #10908, thanks. - -2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Print out the correct filename - (features.conf) in the log message when parkpos options are - incorrect. (closes issue #11295) Reported by: Laureano Patches: - res_features.c.patch uploaded by Laureano (license 265) - - * doc/localchannel.txt: Clarify documentation a bit, include that a - frame has to pass through the core in order for the Local channel - optimization to happen. (closes issue #11246) Reported by: jon - -2007-11-16 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.14 released. - -2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell@digium.com> - - * main/loader.c, include/asterisk/module.h, - build_tools/make_buildopts_h: Temporarily revert revision 89325, - which added md5 magic for keeping track of what build options - were used. We agreed that we should remove this before making a - 1.4 release, and then we can put it back in. Then, we can take a - month or so to play around with it to get it how we want it. - -2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming <kpfleming@digium.com> - - * main/loader.c, include/asterisk/module.h, - build_tools/make_buildopts_h: To help combat problems where - people build external modules (asterisk-addons or others) and - then change the build options of the Asterisk build in a way that - makes the incompatible without warning, this commit introduces an - MD5 signature of the important build-time options and includes - that signature into modules when they are built. When the loader - loads one of these modules and notices the problem, it will emit - a warning to console and refuse to initialize the module, as - doing so could cause the system to be unstable or even crash. If - you upgrade to this version of Asterisk, you must rebuild *all* - of your modules that came from other sources before trying to run - this version. If you are using Digium's G.729 binary codec - module, you will need v33 or newer. - -2007-11-16 15:28 +0000 [r89323] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Make realtime queues accessible from the - QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and - patched by atis, with small modifications from me) - -2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher <tlesher@digium.com> - - * Makefile: Start Asterisk in Debian at a more reasonable time - (since zaptel is at level 20) - - * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found - by valgrind - - * channels/chan_iax2.c: Yet another memory corruption issue. - Reported by: atis Patch by: tilghman Fixes issue #10923 - -2007-11-15 17:19 +0000 [r89296] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Update the SLAStation application to account - for the case where the SLA thread has a call out to the station, - but the user has pressed a line button to answer the call instead - of picking up the handset. If they do, the phone sends out a new - INVITE. So, the SLAStation app must check to see if it is picking - up a ringing trunk, and ensure that the other stations stop - ringing. (reported internally, patched by me, tested by mogorman) - -2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson <mmichelson@digium.com> - - * main/manager.c: Undoing previous commit since I realize it was - wrong - - * main/manager.c: Adding a missing mutex unlock. (closes issue - 11256, reported and patched by ys) - -2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't send re-invites during pending INVITE - transactions. Patch by one47 - thanks! Closes issue #9305 - - * channels/chan_sip.c: Improve support for multipart messages. Code - by gasparz, changes by me (mostly formatting). Thanks, gasparz! - Closes issue #10947 - -2007-11-14 23:23 +0000 [r89275] Tilghman Lesher <tlesher@digium.com> - - * main/app.c: When a recording ends with '#', we are improperly - trimming an extra 200ms from the recording. Reported by: sim - Patch by: tilghman Closes issue #11247 - -2007-11-14 01:15 +0000 [r89260] Joshua Colp <jcolp@digium.com> - - * main/srv.c: Return the proper value when the srv_callback - function executes properly. (closes issue #11240) Reported by: - jtodd - -2007-11-13 21:07 +0000 [r89248-89254] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer - systems which require a third arg to open() when using O_CREAT. - Issue 11238, reported by puzzled. - - * res/res_features.c: Revert change from revision 67064. It is - documented behavior that if a parking extension already exists - while using PARKINGEXTEN, dialplan execution will continue. If - blind transferring to a Park with PARKINGEXTEN, you must keep - this in mind, and handle the failure yourself. Issue 11237, - reported by jon. - -2007-11-13 17:34 +0000 [r89246] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: If we set a value for qualify, we should - actually pay attention to it, instead of overriding the value - -2007-11-13 16:02 +0000 [r89241] Mark Michelson <mmichelson@digium.com> - - * apps/app_mixmonitor.c: Reverting commit made in revision 89205 - since it is unnecessary. Thanks to Kevin for pointing this out - -2007-11-13 13:51 +0000 [r89239] Tilghman Lesher <tlesher@digium.com> - - * main/utils.c: Debugging is running into the 16-lock limit. - Increase to avoid. (This define is only effective when debugging - is turned on, so there's no effect for most installations.) - -2007-11-13 00:56 +0000 [r89205] Mark Michelson <mmichelson@digium.com> - - * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If - only 1 argument is given, then the args.options and - args.post_process strings are uninitialized and could contain - garbage. This change handles this situation properly by only - using arguments that we have parsed. - -2007-11-12 20:46 +0000 [r89194] Jason Parker <jparker@digium.com> - - * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev - -2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher <tlesher@digium.com> - - * main/config.c: If two config writes collide, file corruption - could result. Use a mkstemp() file, instead. Reported by: - paravoid Patch by: tilghman Closes issue #10781 - - * main/channel.c, channels/chan_sip.c: Fix two cases of memory - corruption caused by background threads. Reported by: atis Patch - by: tilghman Fixes issue #10923 - -2007-11-12 11:26 +0000 [r89169-89173] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and - no number was dialed and overlapdial is set, we wait for the ISDN - timeout instead of starting our own timer. added a comment for - the misdn.conf.sample for the overlapdial config option. - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, - channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added - restart all interfaces Restart_Indicator, to automatically send a - RESTART after the L2 of a PTP Port comes up. Also fixed some - places where we have send a RELEASE without need for it. - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a - state/event issue with overlapdial=yes when no extension matched. - removed the general sending of a RELEASE_COMPLETE when we receive - a RELEASE, this is done by mISDNuser/mISDN. This makes it - possible to use asterisk-1.4 with mISDN trunk, but requires users - of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 - (when using the NT mode at all) - - * channels/misdn/isdn_lib.c: fixed the support for CW and therefore - for the reject_cause option. - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - aded ntkeepcalls option, to avoid droÃpping calls when the L2 - goes down on a PTP link. There are some pbx which do turn off the - L1 for a very short while and restart it immediately. normally - T310 should be started and after 10 seconds or so the calls - should be dropped, this is a simple fix wihtout this timer. - -2007-11-08 23:52 +0000 [r89125] Jason Parker <jparker@digium.com> - - * main/say.c: Properly say the seconds here.. Issue 11203, fix - described by vma. - -2007-11-08 21:00 +0000 [r89119] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Rework of the commit I made yesterday to use - the already built-in ast_uri_decode function as opposed to my - home-rolled one. Also added comments. Thanks to oej for pointing - me in the right direction - -2007-11-08 18:45 +0000 [r89115] Jason Parker <jparker@digium.com> - - * configs/res_odbc.conf.sample: Avoid warnings on load when using - sample configuration files. Issue 11195, patch by eliel. - -2007-11-08 16:47 +0000 [r89111] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: I made this same adjustment in trunk to fix - a bug, and it makes sense to do it in 1.4 as well. If an - imapfolder is specified in voicemail.conf, don't ever explicitly - connect to INBOX since it may not exist. - -2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming <kpfleming@digium.com> - - * main/srv.c: fix a glaring bug in the new SRV record handling that - would cause incorrect weight sorting - -2007-11-08 04:55 +0000 [r89103] Tilghman Lesher <tlesher@digium.com> - - * doc/valgrind.txt: Typo - -2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Do not add a sip: to the beginning of the To - URI unless needed. (closes issue #10756) Reported by: goestelecom - - * channels/chan_sip.c: Improve the devicestate logic for multiple - devices. If any are available then the extension is considered - available. (closes issue #10164) Reported by: nic_bellamy - Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic - (license 299) - - * channels/chan_sip.c: Add support for allowing one outgoing - transaction. This means if a response comes back out of order - chan_sip will still handle it. I dream of a chan_sip with real - transaction support. (closes issue #10946) Reported by: flefoll - (closes issue #10915) Reported by: ramonpeek (closes issue #9567) - Reported by: atca_pres - - * channels/chan_sip.c: If callerid is configured in sip.conf use - that for checking the presence of an extension in the dialplan. - (closes issue #11185) Reported by: spditner - -2007-11-07 23:39 +0000 [r89093] Tilghman Lesher <tlesher@digium.com> - - * apps/app_queue.c: The member refcount must be incremented, to - avoid using it after deallocation. A huge thanks go to lvl- for - patiently providing the necessary valgrind output that was - necessary to finding this problem of memory corruption. Reported - by: lvl- Patch by: tilghman Closes issue #11174 - -2007-11-07 22:40 +0000 [r89090] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: This patch makes it possible for SIP phones - to dial extensions defined with '#' characters in extensions.conf - AND maintain their escaped characters when forming URI's (closes - issue #10681, reported by cahen, patched by me, code review by - file) - -2007-11-07 21:40 +0000 [r89088] Steve Murphy <murf@digium.com> - - * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to - 10578, I just ran 1.4 thru valgrind; some of the config leakage - I've already fixed, but it doesn't hurt to double check. I found - and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, - tho. - -2007-11-07 15:56 +0000 [r89085] Mark Michelson <mmichelson@digium.com> - - * main/manager.c: Fixing a segfault in the manager "core show - channels concise" command. (closes issue #11183, reported by arnd - and patched by ys) - -2007-11-07 04:07 +0000 [r89079] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.ael.sample: Suppress AEL warnings on load. - Reported by: eliel Patch by: eliel Closes issue #11178 - -2007-11-06 20:18 +0000 [r89053] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Fix init_classes() so that classes that - actually do have files loaded aren't treated as empty, and - immediately destroyed ... - -2007-11-06 19:09 +0000 [r89046] Jason Parker <jparker@digium.com> - - * codecs/codec_zap.c: Correctly set the total number of channels - from a zaptel transcoder board. SPD-49, patch by Matthew - Nicholson. - -2007-11-06 19:09 +0000 [r89045] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/lock.h: We went to the trouble of creating a - method of tracking failed trylocks, then never turned it on - (oops). - -2007-11-06 18:53 +0000 [r89042] Olle Johansson <oej@edvina.net> - - * main/tdd.c: Bug fixes to tdd support in zaptel. - -2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: If someone were to delete the files used - by an existing MOH class, and then issue a reload, further use of - that class could result in a crash due to dividing by zero. This - set of changes fixes up some places to prevent this from - happening. (closes issue #10948) Reported by: jcomellas Patches: - res_musiconhold_division_by_zero.patch uploaded by jcomellas - (license 282) Additional changes added by me. - -2007-11-06 17:52 +0000 [r89036] Steve Murphy <murf@digium.com> - - * main/config.c: closes issue #8786 - where the [catname](!) and - [catname](othercat1,othercat2,...) notation gets dropped across a - ConfigUpdate (or any other thing that would cause a config file - to be written). While I was at it, I also cleaned up some of the - destroy routines to free up comments, which was not being done. - Made sure the new struct I introduced is also cleaned up properly - at destruction time. My code handles multiple template - inclusions. Many thanks to ssokol for his patch, which, while not - literally used in the final merge, served as a foundation for the - fix. - -2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make it so that if a peer is determined to - be unreachable using qualify their devicestate will report back - unavailable. (closes issue #11006) Reported by: pj - - * channels/chan_zap.c: Fix improbable but possible memory leaks in - chan_zap. (closes issue #11166) Reported by: eliel Patches: - chan_zap.c.patch uploaded by eliel (license 64) - -2007-11-06 13:50 +0000 [r88931] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: Remove some checks to see if locks are - initialized from the non-DEBUG_THREADS versions of the lock - routines. These are incorrect for a number of reasons: - It - breaks the build on mac. - If there is a problem with locks not - getting initialized, then the proper fix is to find that place - and fix the code so that it does get initialized. - If additional - debug code is needed to help find the problem areas, then this - type of things should _only_ be put in the DEBUG_THREADS - wrappers. - -2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/srv.h: update comment to match the state of the - code - -2007-11-05 23:29 +0000 [r88826] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: Reworked deadlock avoidance in __ast_read. - Restored audio to callback agents. (closes issue #11071, reported - by callguy, patched by me, tested by callguy and Ted Brown) - -2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant <russell@digium.com> - - * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related - to channel variables, I went looking around at the ways that - channel variables are handled. In general, they were not handled - in a thread-safe way. The channel _must_ be locked when reading - or writing from/to the channel variable list. What I have done to - improve this situation is to make pbx_builtin_setvar_helper() and - friends lock the channel when doing their thing. Asterisk API - calls almost all lock the channel for you as necessary, but this - family of functions did not. (closes issue #10923, reported by - atis) (closes issue #11159, reported by 850t) - - * channels/chan_sip.c: When traversing the list of channel - variables here in transmit_invite(), the asterisk channel must be - locked, as this data may change at any time. (I have seen - numerous reports of crashes related to the handling of channel - variables. There are a couple of issues on the bug tracker - related to it, but it has also been noted on IRC and mailing - lists. So, I am finding and fixing some places where channel - variables are handled improperly.) - - * channels/chan_sip.c: Fix up some indentation. - - * main/srv.c, include/asterisk/srv.h: Merge changes from - asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV - record support in Asterisk was broken. There was no guarantee on - what record Asterisk would choose to actually use. This set of - changes improves the situation by ensuring that Asterisk will - choose the highest priority record. - - * main/channel.c: Merge the last bit of changes from - asterisk/team/russell/readq-1.4 The issue here is that the - channel frame readq handling got broken when the code was - converted to use the linked list macros. It caused corruption of - the list head and tail pointers. So, I fixed up the usage of the - linked list macros and in passing, simplified the code. I also - documented what the code is doing, as it was a bit difficult to - figure out at first. This bug showed itself with crashes showing - messed up head/tail pointers for the readq. However, there are a - couple of crashes that aren't quite as obvious, but I think may - be related. So, if your bug gets closed by this commit, but you - still have a problem, please reopen or create a new bug report. - (closes issue #10936) (closes issue #10595) (closes issue #10368) - (closes issue #11084) (closes issue #10040) (closes issue #10840) - -2007-11-05 18:47 +0000 [r88671] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: If a SIP channel is put on hold multiple - times do not keep incrementing the onHold value. (closes issue - #11085) Reported by: francesco_r Tested by: blitzrage (closes - issue #10474) Reported by: acennami - -2007-11-05 17:46 +0000 [r88624] Russell Bryant <russell@digium.com> - - * main/channel.c: Fix up datastore handling in ast_do_masquerade(). - The code is intended to move any channel datastores from the old - channel to the new one. However, it did not use the linked list - macros properly to accomplish the task. The existing code would - only work if there was only a single datastore on the old - channel. - -2007-11-05 17:19 +0000 [r88585] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Make sure we destroy the config structure on - configuration failure. Issue 11163, patch by eliel. - -2007-11-05 16:20 +0000 [r88539] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c: Don't check used pooled connections for - connection status, as it will cause issues for prepared queries. - Reported by: Nick Gorham (via -dev list) Patch by: tilghman - -2007-11-04 22:38 +0000 [r88471] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/stringfields.h, main/channel.c, - apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c: - Rename ast_string_field_free_pool to - ast_string_field_free_memory, and ast_string_field_free_all to - ast_string_field_reset_all to avoid misuse (due to too similar - names and an error in documentation). Fix two related memory - leaks in app_meetme. No need to merge to trunk, different fix - already applied there. Not applicable to 1.2 - -2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make subscribecontext behave as advertised. - It will now look for the presence of a hint in the given context - (be it subscribecontext or context). (closes issue #10702) - Reported by: slavon - - * channels/chan_sip.c: If an INFO request within a dialog is - received with a content length of 0 simply send back a 200 OK. It - is valid to do this and the remote side is probably using it to - make sure the signalling is still alive. (closes issue #5747) - Reported by: chandi Patches: infofix-81430-1.patch uploaded by - IgorG (license 20) - -2007-11-02 16:51 +0000 [r88283] Jason Parker <jparker@digium.com> - - * main/say.c: We need to make sure to specify a language to - ast_fileexists, otherwise it may fail for anything besides en - Issue 11147, fix discovered by both citats and myself - (independently), with input from Corydon76 - -2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy - Patch by: ys Closes issue #11143 - - * doc/valgrind.txt (added): Add some notes on using valgrind - -2007-11-01 16:21 +0000 [r88078] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Make sure we set the poll fds to NULL after - free()ing it. Part of issue 11017, patch by tzafrir. - -2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Fix up commit for my Zap channel with spies in - Meetme fix. (thanks Tony Mountifield!) - - * apps/app_meetme.c: If a Zap channel contains a spy or a spy is - added take it out of the conference in kernel space and make it - go through Asterisk so the spy gets audio from both sides. - (closes issue #10060) Reported by: mparker - -2007-10-31 21:23 +0000 [r87906-87908] Jason Parker <jparker@digium.com> - - * res/res_jabber.c: Make sure we free some allocated memory before - returning. Issue 11131, patch by eliel. - - * channels/chan_gtalk.c: Don't try to allocate memory that we're - just going to re-allocate later anyways. Issue 11130, patch by - eliel. - -2007-10-31 18:03 +0000 [r87852] Tilghman Lesher <tlesher@digium.com> - - * Makefile: Create samples for ALL of the available options in - asterisk.conf - -2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy <murf@digium.com> - - * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan - save' cli command saves a file where the semicolon is not - escaped. Fixed this; User also wanted comments to be preserved - across dialplan save, but this is impossible at this point in - time, because comments are not stored in the dialplan. They are - 'compiled' out of extensions.conf. The only way to preserve those - comments is to use the config file reader/writer that the GUI - uses to allow online user edits. extensions.conf is first and - foremost, a config file, and is read in by the normal config-file - reading routines. Then, it is processed into a dialplan - (context/exten structs). - - * pbx/pbx_ael.c: Included some verbage in the check_includes func, - to inform the user that included contexts that have no match in - the AEL, might be OK, as AEL cannot check in the extensions.conf - or the in-memory contexts, as they may not be there at the time - of the check. - -2007-10-30 23:02 +0000 [r87739] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD - Reported by: ys Fixed by: ys Closes issue #11116 - -2007-10-30 21:19 +0000 [r87686] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Merge the changes from - team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a - race condition related to the handling of POKEing peers. - Essentially, a reference to a peer is held by the scheduler when - there are pending callbacks, but the reference count didn't - reflect it. So, it was possible for a peer to hit a reference - count of zero and have its destructor begin to be called at the - same time that the scheduler thread ran a POKE related callback. - If that happened, a crash would likely occur. (closes issue - #11082, closes issue #11094) - -2007-10-30 20:29 +0000 [r87650] Jason Parker <jparker@digium.com> - - * channels/Makefile: Only try to clean out h323/ if the - h323/Makefile exists. - -2007-10-30 16:13 +0000 [r87571] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Add two more checks before printing out a - warning message about bridging. If either channel has hungup of - course the bridge will have failed. (closes issue #10009) - Reported by: dimas - -2007-10-30 15:45 +0000 [r87567] Jason Parker <jparker@digium.com> - - * main/editline/np/vis.c: Fix build of editline on Solaris. Issue - 11113, patch by snuffy. - -2007-10-30 15:10 +0000 [r87534] Joshua Colp <jcolp@digium.com> - - * apps/app_followme.c: Return 1.4 to a state where it builds. - Changing the arguments to a function and not changing where they - are used is bad, mmmk? - -2007-10-30 14:31 +0000 [r87514] BJ Weschke <bweschke@btwtech.com> - - * apps/app_followme.c: Fix issue where the recorded name wasn't - getting removed correctly. (closes issue #11115) Reported by: - davevg Patches: followme-v3.diff - -2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/gsm: missed one directory - - * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs, - codecs/lpc10, main/db1-ast, main/editline, main, - codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile, - codecs/lpc10/Makefile, utils, codecs, agi, - main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr: - clean up (and ignore) assembler and preprocessor intermediate - files if any are created during the build - - * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is - already there (used when debugging preprocessor issues) because - the compiler will whine about each compile command - -2007-10-29 21:06 +0000 [r87427] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removing a completely unnecessary quota - check from IMAP code. - -2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant <russell@digium.com> - - * main/utils.c, include/asterisk/lock.h: Add some more details to - the output of "core show locks". When a thread is waiting for a - lock, this will now show the details about who currently has it - locked. (inspired by issue #11100) - - * main/astmm.c: Remove a lock that doesn't make any sense. The - regions lock needs to be held when traversing the list of - allocated chunks so that they can be printed out to the CLI. - (Thanks to eliel on #asterisk-dev for pointing this out!) - -2007-10-29 17:20 +0000 [r87342] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix issue where if both sides of the dialog - cancelled the dialog at the same time chan_sip could kepe - retransmitting a response for no reason. (closes issue #9566) - Reported by: atca_pres Patches: bug9566.patch uploaded by oej - -2007-10-29 17:13 +0000 [r87340] Jason Parker <jparker@digium.com> - - * funcs/func_realtime.c, funcs/func_cut.c: Allow some function - modules to compile under dev mode. Issue 11104, patch by andrew. - -2007-10-29 14:23 +0000 [r87294] Joshua Colp <jcolp@digium.com> - - * main/utils.c: Fix issue with ast_unescape_semicolon going into an - endless loop. (closes issue #10550) Reported by: ramonpeek - Patches: unescape-85177-1.patch uploaded by IgorG (license 20) - -2007-10-28 13:46 +0000 [r87262] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c, - funcs/func_cut.c: Add autoservice to several more functions which - might delay in their responses. Also, make sure that func_odbc - functions have a channel on which to set variables. Reported by - russell Fixed by tilghman Closes issue #11099 - -2007-10-26 16:34 +0000 [r87168] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c, - pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c, - include/asterisk/ael_structs.h, pbx/ael/ael.tab.h, - utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16, - pbx/ael/ael.flex: closes issue #11086 where a user complains that - references to following contexts report a problem; The problem - was REALLy that he was referring to empty contexts, which were - being ignored. Reporter stated that empty contexts should be OK. - I checked it out against extensions.conf, and sure enough, empty - contexts ARE ok. So, I removed the restriction from AEL. This, - though, highlighted a problem with multiple contexts of the same - name. This should be OK, also. So, I added the extend keyword to - AEL, and it can preceed the 'context' keyword (mixed with - 'abstract', if nec.). This will turn off the warnings in AEL if - the same context name is used 2 or more times. Also, I now call - ast_context_find_or_create for contexts now, instead of just - ast_context_create; I did this because pbx_config does this. The - 'extend' keyword thus becomes a statement of intent. AEL can now - duplicate the behavior of pbx_config, - -2007-10-26 13:54 +0000 [r87120] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_curl.c: The addition of autoservice to func_curl - additionally made func_curl dependent on the existence of a - channel, with no real reason. This should make func_curl once - again work without a channel. Reported by jmls. Fixed by - tilghman. Closes issue #11090 - -2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, include/asterisk/linkedlists.h: appending one - list to another should leave the first list empty, and not - require the user to do that - -2007-10-25 22:53 +0000 [r87067] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_cut.c: Backport alternate encoding of newline - delimiters from trunk to 1.4, as approved by Russell Reported by - blitzrage Closes issue #10903 - -2007-10-24 20:56 +0000 [r86982] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Correctly respect hidecalleridname - configuration option. Simplify code slightly in the process. - Issue 11079, reported by ddv2005 - -2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy <murf@digium.com> - - * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable - to specify app:spec in hint arguments - - * funcs/func_logic.c: closes issue #11052 -- where nothing after - the ? will allow un-initialized variable values to corrupt and - crash asterisk on 64-bit platforms - - * main/Makefile: this update to Makefile corrects how ast_expr2f.c - should be generated - - * main/ast_expr2f.c: This should get rid of a really, really - irritating warning generated by some 64-bit platforms from libc, - where free(0) is frowned upon - -2007-10-22 21:36 +0000 [r86836] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: If lock tracking is not enabled, then we - can not attempt to log any mutex failures. If so, we could end up - in infinite recursion. The only lock that is affected by this is - a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes - issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys - (license 281) - -2007-10-22 17:38 +0000 [r86787] Tilghman Lesher <tlesher@digium.com> - - * main/astmm.c: Minor FreeBSD build fix - -2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: After reading online I have confirmed that - Record-Route headers should be copied to 1xx responses as well. - (closes issue #10113) Reported by: makoto - - * apps/app_controlplayback.c: Make sure res is a positive value - before performing the check to determine whether the user stopped - it or not. (closes issue #11023) Reported by: cfc - -2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant <russell@digium.com> - - * main/channel.c: Don't leak a frame in the case that an END frame - is received and the time since the BEGIN is less than that of the - defined minimum DTMF duration. (closes issue #11051) Reported by: - casper Patches: channel.c.86664.diff uploaded by casper (license - 55) - - * include/asterisk/lock.h: Update the static mutex initializer to - include the initialization of the internal mutex used to protect - the lock debugging data. (closes issue #11044, patch suggested by - Ivan) - -2007-10-22 14:48 +0000 [r86694] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Account for the fact that sometimes headers - may be terminated with \r\n instead of just \n (closes issue - #11043, reported by yehavi) - -2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Move log message to before the frame it - references is freed. (closes issue #11050) Reported by: slavon - Patches: channel.c.86662.diff uploaded by casper (license 55) - - * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes - issue #11041) Reported by: jsmith Patches: - asterisk-dundicomplete.diff.txt uploaded by jamesgolovich - (license 176) - - * main/loader.c: Fixes for building under OpenSolaris. (closes - issue #11047) Reported by: snuffy Patches: 11047-fixes.diff - uploaded by snuffy (license 35) - -2007-10-22 09:21 +0000 [r86598] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send - DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan - does not match after an overlap call. Also added out_cause=1 - -2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp <jcolp@digium.com> - - * main/app.c: When returning a DTMF digit from - ast_control_streamfile cast it as a char so that 0 does not - overlap with the success return code. (closes issue #11023) - Reported by: cfc - - * channels/chan_sip.c: Fix two issues with domains and transfers. - If a port was given in the hostname it was treated as part of the - hostname. If domains were configured but external domains were - not enabled all transfers would be considered remote. (closes - issue #11027) Reported by: ramonpeek Patches: 11027-1.diff - uploaded by ramonpeek (license 266) - - * channels/chan_sip.c: Set port number in received as information - for registrations as well. (closes issue #11028) Reported by: - brad-x - -2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development <support@transnexus.com> - - * apps/app_osplookup.c: Fixed OSP module did not report - source/devinfo IP in correct format. - -2007-10-18 22:01 +0000 [r86405-86406] Jason Parker <jparker@digium.com> - - * Makefile: Correct documentation. I removed the wrong line.. - - * Makefile: Add documentation for options in asterisk.conf Issue - 11029, patch by eserra - -2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant <russell@digium.com> - - * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous - commit. - - * configs/iax.conf.sample, channels/chan_iax2.c: Add support for - setting the maximum trunk size for IAX2 trunking - - * main/channel.c, include/asterisk/channel.h: The channel needs to - stay locked while running timer callbacks, as they access and - modify channel data that may change elsewhere. I went through - every timer callback in the source tree to make sure that none of - them did any additional locking that could introduce deadlocks, - and all is well. (closes issue #10765) Reported by: Ivan Patches: - ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license - 229) - -2007-10-18 17:38 +0000 [r86328] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: If a non-existent file is specified to be - played either as a periodic announcement or as a hold/position - announcement, the caller would be kicked out of the queue. No - longer does this happen. - -2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant <russell@digium.com> - - * codecs/codec_zap.c: Execute the RELEASE operation on transcoder - channels in the destroy callback. (patch from jsloan) - - * main/utils.c: Revert a change that I made for issue #10979 which, - as has been pointed out to me in issue #11018, doesn't really - make sense. There is no reason to have the base64 decode function - force a '\0' terminated buffer, when the result is almost always - binary, anyway. In fact, this caused some breakage, as some code - in res_crypto passed in a buffer exactly the right size to get - its binary result, which got stomped on by this patch. (closes - issue #11018, reported by dimas) - -2007-10-17 21:39 +0000 [r86202] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Changing the strategy field of the call_queue - struct to be signed instead of unsigned, since the code attempts - to set the strategy to -1 if you specify a bogus strategy. While - this isn't a huge issue in 1.4, it could be a problem for someone - who, say, tries to use the roundrobin strategy in trunk (despite - all the deprecation warnings in 1.4). - -2007-10-17 17:57 +0000 [r86149] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: If Asterisk is in the middle of shutting - down, respond to OPTIONS with 503 Unavailable. (closes issue - #10994) Reported by: eserra Patches: sip-options-503.patch - uploaded by eserra (license 45) - -2007-10-17 16:58 +0000 [r86117] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Whoops, forgot to remove the original - sip_scheddestroy. (closes issue #11010) Reported by: vadim - -2007-10-17 15:23 +0000 [r86066] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: When runuser/rungroup is specified, a remote - console could only be attained by root (Closes issue #9999) - -2007-10-17 15:06 +0000 [r86063] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't schedule dialog destruction if a - MESSAGE is received using an existing dialog. (closes issue - #11010) Reported by: vadim - -2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample: Since monitor-join is deprecated now, - remove the example from the sample queues.conf file - - * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of - the monitor-join queue option - - * apps/app_queue.c: Adding deprecated warning to monitor-join - option, since the plan is to no longer support this in favor of - monitor-type = mixmonitor (related to issue #10885) - -2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: really picky formatting tweak ... - - * include/asterisk/lock.h: Some locking errors exposed the fact - that the lock debugging code itself was not thread safe. How - ironic! Anyway, these changes ensure that the code that is - accessing the lock debugging data is thread-safe. Many thanks to - Ivan for finding and fixing the core issue here, and also thanks - to those that tested the patch and provided test results. (closes - issue #10571) (closes issue #10886) (closes issue #10875) (might - close some others, as well ...) Patches: (from issue #10571) - ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license - 229) - a few small changes by me - -2007-10-16 21:14 +0000 [r85958] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Trying to remove a non-dynamic queue member via - dynamic means can lead to some interesting (read nasty) - situations. This patch clears up the issue by making only dynamic - queue members removable via dynamic methods. - -2007-10-16 19:41 +0000 [r85921] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/localtime.c: Also set up gmtoff (this is used in the - %z gnu extension to strftime) Reported and fixed by jcmoore - Closes issue #11002 - -2007-10-16 19:10 +0000 [r85896] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Remove a pointless lock. - -2007-10-16 15:21 +0000 [r85852] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixing a double free which happens in the - statechange thread. (closes issue #10987, reported by andrew) - -2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp <jcolp@digium.com> - - * apps/app_hasnewvoicemail.c: Check to make sure a value has been - given to the VMCOUNT dialplan function. (closes issue #10996) - Reported by: marsosa - - * main/threadstorage.c: Fix memory allocation issue in - threadstorage. (closes issue #10995) Reported by: snuffy Patches: - new-patch.diff uploaded by snuffy (license 35) - -2007-10-16 10:46 +0000 [r85800] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Fix the output for this channel help CLI - command - -2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Ensure that no pending state changes are leaked - when the device state change thread gets stopped on module - unload. - - * apps/app_queue.c: Previously, app_queue created a thread to - handle every single device state change. I changed this a while - ago in trunk for performance reasons. However, bug 8407 points - out that it is actually a race condition, causing device state - changes to get processed in random order. So, I backported my - changes from trunk to 1.4. (closes issue #8407, patch provided by - tim_ringenbach, committed patch by me) - -2007-10-15 20:29 +0000 [r85687] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Don't execute a gosub if the arguments is - zero-len (not just NULL) Reported by davevg Fixed by me Closes - issue #10985 - -2007-10-15 20:21 +0000 [r85686] Russell Bryant <russell@digium.com> - - * main/say.c: Add a small fix for the tw version of saying dates. - (closes issue #7827) Reported by: sharkey Patches: say.nits.patch - uploaded by sharkey (license 172) - -2007-10-15 20:15 +0000 [r85684] Jason Parker <jparker@digium.com> - - * Makefile: Properly use DESTDIR in 'config' target. Do not try to - run chkconfig or similar if using DESTDIR. Issue 10938, patch by - cabal95. - -2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant <russell@digium.com> - - * main/utils.c: Be pedantic about handling memory allocation - failure. - - * main/utils.c: The loop in the handler for the "core show locks" - could potentially block for some amount of time. Be a little bit - more careful and prepare all of the output in an intermediary - buffer while holding a global resource. Then, after releasing it, - send the output to ast_cli(). - - * channels/chan_sip.c: Make the default for the srvlookup option to - be yes. It doesn't really make sense for it to default to off. - The default configuration file has it on, and proper RFC - behavior, as indicated by a comment in the code, is for it to be - on. So, let's have it on by default to make lives easier. (closes - issue #10954, suggested by jtodd) - -2007-10-15 16:39 +0000 [r85571] Joshua Colp <jcolp@digium.com> - - * configs/features.conf.sample: Document that DTMF based features - only work when two channels are bridged together. (closes issue - #10773) Reported by: pbayley - -2007-10-15 16:34 +0000 [r85561] Russell Bryant <russell@digium.com> - - * include/asterisk/strings.h: Make a few changes so that characters - in the upper half of the ISO-8859-1 character set don't get - stripped when reading configuration. (closes issue #10982, - dandre) - -2007-10-15 16:22 +0000 [r85559] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Bring both DTMF begin and end frames up through to - the core for DTMF feature handling. (closes issue #10826) - Reported by: dimas - -2007-10-15 15:40 +0000 [r85556] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Ensure the buffer passed to - ast_canmatch_extension() is properly initialized so that it is - null terminated. (issue #10977) Reported by: dimas Patches: - pbxdundi.patch uploaded by dimas (license 88) - small mods by me - -2007-10-15 14:55 +0000 [r85552] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: If Monitor or a spy was added to a P2P or native - bridged channel bring the channel back to the generic bridging - core so the monitor or spy operations work. (closes issue #10943) - Reported by: julianjm - -2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant <russell@digium.com> - - * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled. - (closes issue #10980) Reported by: casper Patches: - db.c.84633.diff uploaded by casper (license 55) - - * main/asterisk.c: Make sure remote consoles unmute themselves - again after reconnecting. (closes issue #10847) Reported by: atis - Patches: console_unmute_on_reconnect.patch uploaded by atis - (license 242) - - * main/utils.c: Make sure that the base64 decoder returns a - terminated string. (closes issue #10979) Reported by: ys Patches: - util.c.diff uploaded by ys (license 281) - small mods by me - - * pbx/pbx_config.c: Don't create the context for users in - users.conf until we know at least one user exists. (closes issue - #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by - dimas (license 88) - -2007-10-13 15:26 +0000 [r85536] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.ael.sample: Remove deprecated syntax from - sample ael file Reported and patched by: dimas Closes issue - #10967 - -2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant <russell@digium.com> - - * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an - issue with console verbosity when running asterisk -rx to execute - a command and retrieve its output. The issue was that there was - no way for the main Asterisk process to know that the remote - console was connecting in the -rx mode. The way that James has - fixed this is to have all remote consoles muted by default. Then, - regular remote consoles automatically execute a CLI command to - unmute themselves when they first start up. (closes issue #10847) - Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded - by jamesgolovich (license 176) - - * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly - handle the case where read() may return the text for more than - one CLI command at once for a remote console. (closes issue - #10888) Reported by: jamesgolovich Patches: - asterisk-climultiple.diff.txt uploaded by jamesgolovich (license - 176) - -2007-10-12 18:30 +0000 [r85523] Tilghman Lesher <tlesher@digium.com> - - * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address - -2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant <russell@digium.com> - - * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not - SDMI. (closes issue #10959) - - * pbx/pbx_realtime.c: Fix the potential use of an uninitialized - buffer in a log message. (closes issue #10958) Reported by: dimas - Patches: realtime.patch uploaded by dimas (license 88) - -2007-10-11 15:26 +0000 [r85397] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: When creating a new packet don't try to stop - retransmission of it. It was just allocated/created so it's - impossible for it to have already been scheduled. (closes issue - #10945) Reported by: flefoll Patches: - chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll - (license 244) - -2007-10-11 04:35 +0000 [r85356] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: A dollar sign by itself, not indicating a start of a - variable or expression prematurely ends substitution (closes - issue #10939) - -2007-10-10 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.13 released. - -2007-10-10 15:56 +0000 [r85316] Russell Bryant <russell@digium.com> - - * include/asterisk/file.h: I introduced a new member to the - ast_filestream struct in 1.4.12, but put it in the middle of the - struct, instead of at the end. One of the Debian folks, paravoid, - pointed out that this breaks binary compatability with modules - compiled against older headers. So, I'm moving the new member to - the end of the struct to resolve the situation. - -2007-10-10 15:51 +0000 [r85315] Mark Michelson <mmichelson@digium.com> - - * main/utils.c: The thread ID should be unsigned. - -2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: If devicestate is passed a port number strip - it out. (closes issue #10930) Reported by: ibc - - * channels/chan_sip.c: Add support for handling a 182 Queued - response. (closes issue #10924) Reported by: ramonpeek Patches: - queued-182.diff uploaded by ramonpeek (license 266) - -2007-10-10 14:26 +0000 [r85276] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: A bunch of changes from sprintf to - snprintf. See security advisory AST-2002-022 - -2007-10-10 14:14 +0000 [r85242] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Close voicemail message description file if - duration did not meet the minimum, or else we will eventually run - out of file descriptors. (closes issue #10918) Reported by: - brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license - 279) - -2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/frame.h: use a macro instead of an inline - function, so that backtraces will report the caller of - ast_frame_free() properly - -2007-10-09 21:55 +0000 [r85158] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, main/utils.c, include/asterisk/lock.h: This - commit fixes the following issues: - Deadlock in ast_write (issue - #10406) - Deadlock in ast_read (issue #10406) - Possible mutex - initialization error in lock.h (issue #10571) - -2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't perform a reinvite if a transfer is in - progress. (issue #10915) Reported by: ramonpeek - - * main/rtp.c: Only update codec information if the channel has a - technology private structure. (issue #10915) Reported by: - ramonpeek - - * main/rtp.c: Update codec information as well as address when - doing hold reinvites. (issue #10868) Reported by: mavince - - * main/channel.c: Don't keep trying to native bridge if either of - the channels are involved in a masquerade operation to be done. - (closes issue #10696) Reported by: tbelder - -2007-10-08 03:28 +0000 [r84957] Russell Bryant <russell@digium.com> - - * Makefile.rules: Enable file dependency tracking for _all_ builds, - and not just for builds with dev-mode enabled. I have seen enough - problems caused by this that I don't think it's worth keeping. I - want to continue to encourage anybody that is interested to - continue to run Asterisk from svn. Furthermore, I do not want - their systems to break when we change a structure definition in a - header file. :) - -2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Presence packets from a client who's connected - with our Jabber ID are valid, therefore, those clients must be - considered as buddies. The resource string helps us make the - distinction between clients. Closes issue #10707, reported by - yusufmotiwala. - - * res/res_jabber.c: Prevent Asterisk from crashing when receiving a - presence packet without resource from a buddy that is known to - have a resource list. Revert a change I previously made, where - Asterisk could point to a freed memory location. - -2007-10-05 19:42 +0000 [r84851] Tilghman Lesher <tlesher@digium.com> - - * main/db.c: Log exactly why we can't open the database, if we fail - (closes issue #10887) - -2007-10-05 18:55 +0000 [r84818] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Update the remembered RTP peer information when - putting an endpoint on hold or taking it off hold so that the RTP - stack does not initiate a needless reinvite. (closes issue - #10868) Reported by: mavince - -2007-10-05 16:44 +0000 [r84783] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Do deadlock avoidance in a couple more - places. You can't lock two channels at the same time without - doing extra work to make sure it succeeds. (closes issue #10895, - patch by me) - -2007-10-05 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.12.1 released. (This is mainly to include the - app_queue fix for a memory leak on reload, but includes a couple - of other bug fixes, as well.) - -2007-10-05 01:39 +0000 [r84742] Russell Bryant <russell@digium.com> - - * main/manager.c: Fix a copy/paste error in the description of - UpdateConfig that was pointed out by JerJer on #asterisk-dev - -2007-10-04 21:57 +0000 [r84692] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Don't allocate space for queue members unless - it's needed. You end up deleting dynamic members on a reload. Not - good. closes issue (#10879, reported by dazza76, patched by me) - -2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: callers of sig2str already add the word - 'signalling' in the appropriate place, so don't duplicate it - -2007-10-04 14:51 +0000 [r84637] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Create a duplicate of the channel's member name - as the tab completion stuff will free it. (closes issue #10884) - Reported by: adamg - -2007-10-03 22:59 +0000 [r84581] Tilghman Lesher <tlesher@digium.com> - - * main/rtp.c: When an RFC 2833 event is sent that we don't - recognize, ignore it, don't queue a NULL digit (closes issue - #10877) - -2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call - in a switch expr doesn't execute correctly, because the commas in - the function args are not converted to vertbars before the func - is called. I modified just the switch code to convert the commas - to vertbars if there, but if more of these sort of probs are - found, I may have to resort to something a little more - fundamental. We'll see, I guess. - - * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ref.ael-vtest17, - pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, - pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a - null input to a switch statement results in a hangup; since - switch is implemented with extensions, and the default case is - implemented with a '.', and the '.' matches 1 or more remaining - characters, the case where 0 characters exist isn't matched, and - the extension isn't matched, and the goto fails, and a hangup - occurs. Now, when a default case is generated, it also generates - a single fixed extension that will match a null input. That - extension just does a goto to the default extension for that - switch. I played with an alternate solution, where I just tack an - extra char onto all the patterns and the goto, but not the - default case's pattern. Then even a null input will still have at - least one char in it. But it made me nervous, having that extra - char in , even if that's a pretty secret and low-level issue. - -2007-10-02 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.12 released. - -2007-10-02 20:06 +0000 [r84474] Russell Bryant <russell@digium.com> - - * Makefile, build_tools/prep_tarball: * Don't build the - menuselect-tree for the tarball, as it requires running the - configure script first * Change the Makefile to note that - menuselect-tree depends on the configure script. - -2007-10-02 19:01 +0000 [r84410-84437] Jason Parker <jparker@digium.com> - - * res/res_features.c: Fix some odd formatting I missed.. - - * res/res_features.c: Finish up on transferee channel before return - on failure. Issue 10821, patch by Ivan - -2007-10-02 14:12 +0000 [r84370] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Use snprintf instead of sprintf in one - place. There is no vulnerability here due to various buffer sizes - around the code, but I still didn't like seeing a non - length-limited copy of data coming off of the wire into a stack - buffer, as this would be a problem in the future if buffer sizes - elsewhere got changed or size limitations removed ... - -2007-10-02 09:48 +0000 [r84345] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: terminate USERUSER String with 0 - -2007-10-01 21:52 +0000 [r84291] Jason Parker <jparker@digium.com> - - * Makefile, Makefile.rules, channels/Makefile: Add dist-clean - support for subdirs. Change h323 to only remove the Makefile on a - dist-clean, rather than a clean. This fixes a bug I found with - trying to run make after a make clean - -2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/channel.c, main/manager.c, channels/chan_agent.c: moved - get_base_channel() code from action_redirect to - ast_channel_masquerade() for issue 7706 and BE-160 - -2007-10-01 21:18 +0000 [r84273] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy... - -2007-10-01 21:07 +0000 [r84271] Russell Bryant <russell@digium.com> - - * main/utils.c, include/asterisk/lock.h: Fulfull a feature request - from Qwell on the "core show locks" output. It will now note the - lock type for each lock that a thread holds. (mutex, rdlock, or - wrlock) - -2007-10-01 20:27 +0000 [r84239] Steve Murphy <murf@digium.com> - - * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue - #10777 -- by returning a null for the parse tree when there's - really nothing there, and making sure we don't try to do checking - on a null tree. - -2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant <russell@digium.com> - - * res/res_agi.c: Add another sanity check in the AGI read loop. We - really don't care about EAGAIN unless we didn't read an entire - line. If there is a newline at the end if the read buffer, break, - because we got the whole thing. (reported and patched by bmd) - - * include/asterisk/lock.h: Show rwlocks in the "core show locks" - output. Before, it only showed mutexes. - - * channels/Makefile: Remove another file in "make clean". (closes - issue #10814, paravoid) - - * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit. - -2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp <jcolp@digium.com> - - * configs/usbradio.conf.sample (removed): Remove chan_usbradio - config file from tree, it is not present in here. (closes issue - #10839) Reported by: casper - - * res/res_musiconhold.c: Fix randomness. save_pos was being set to - 0 initially instead of -1, causing it to jump to position 0 when - moh started. (closes issue #10859) Reported by: jamesgolovich - Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich - (license 176) - - * apps/app_dial.c: Only attempt early bridging if the options given - to Dial() permit it. (closes issue #10861) Reported by: peekyb - -2007-09-30 20:02 +0000 [r84146] Russell Bryant <russell@digium.com> - - * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++ - modules. The load and reload parameters were in the wrong place. - (closes issue #10846, alebm) - -2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added), - pbx/ael/ael-test/ael-ntest22/t1/b.ael (added), - pbx/ael/ael-test/ael-ntest22/t1/c.ael (added), - pbx/ael/ael-test/ael-ntest22/t2/d.ael (added), - pbx/ael/ael-test/ael-ntest22/t2/e.ael (added), - pbx/ael/ael-test/ael-ntest22/t2/f.ael (added), - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22 - (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added), - pbx/ael/ael-test/ref.ael-test3, - pbx/ael/ael-test/ael-ntest22/t3/h.ael (added), - pbx/ael/ael-test/ref.ael-test4, - pbx/ael/ael-test/ael-ntest22/t3/i.ael (added), - pbx/ael/ael-test/ael-ntest22/t3/j.ael (added), - pbx/ael/ael-test/ael-ntest22/qq.ael (added), - pbx/ael/ael-test/ael-ntest22/t1 (added), - pbx/ael/ael-test/ael-ntest22/t2 (added), - pbx/ael/ael-test/ael-ntest22/t3 (added), - pbx/ael/ael-test/ael-ntest22/extensions.ael (added), - pbx/ael/ael-test/ael-ntest22 (added): This is a regression update - that matches what I did in 84134 for AEL regressions. - - * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes - 10786; All config files support #include with globbing (you know, - *,[chars],?,{list,list},etc), so I've updated the AEL system to - support this also. - -2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher <tlesher@digium.com> - - * main/say.c: Correct pronunciations of numbers for .nl (Closes - issue #10837) - - * main/channel.c: Avoid a deadlock with ALL of the locks in the - masquerade function, not just the pairs of channels. (Closes - issue #10406) - -2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/manager.c, channels/chan_agent.c, - include/asterisk/channel.h: if an Agent is redirected, the base - channel should actually be redirected. This was causing multiple - issues, especially issue 7706 and BE-160 - -2007-09-27 00:01 +0000 [r83976] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: remove a todo item that has been completed - -2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_alsa.c: avoid the weird usage of assert() in the - ALSA header files that gcc 4.2 wants to complain about - -2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: I changed my mind ... I think this should be - a LOG_NOTICE. - - * channels/chan_sip.c: Add a log message that was requested by the - masses in the developer tutorial session at Astricon. chan_sip - did not output any message when a call was rejected because the - extension was not found. This adds a verbose message (at verbose - level 3) to note when this happens. - - * channels/chan_misdn.c: Fix building chan_misdn under dev-mode. - (please run the configure script with --enable-dev-mode so this - doesn't happen again ...) - -2007-09-26 18:35 +0000 [r83879] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c: Remove unused 4k of memory on the program - stack (closes issue #10827) - -2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher <tlesher@digium.com> - - * main/app.c: jmls pointed out that unsetting the group and setting - the group to the blank string aren't quite the same. - - * build_tools/make_defaults_h: In the source, keys are relative to - the datadir, not varlib (which is the same in most cases, but - it's good to be accurate). Closes issue #10811 - - * doc/realtime.txt: Oops. Removed the unworkable workaround. This - note should never have been in the release. - - * main/app.c: Making change to group splitting, as discussed on the - -dev list. The main effect of this will be to permit - Set(GROUP([cat])=), i.e. unsetting a group. - -2007-09-24 07:54 +0000 [r83620] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed round_robin group dial method, this - never worked well on BRI Ports (2 channels) - -2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes - are made here for the first arg in the for(arg1; arg2; arg3) {} - statement, as were done for the 3rd arg. It can now be an - assignment that will embedded in a Set() app, or a macro call, or - an app call. - - * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for - statement is now wrapped in Set() only if there's an '=' in that - string. Otherwise, if it begins with '&', then a Macro call is - generated; otherwise it is made into an app call. A bit more - accomodating, keeps the new guys happy, and the guys with ael-1 - code should be happy, too - -2007-09-21 14:37 +0000 [r83432] Russell Bryant <russell@digium.com> - - * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c, - channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c, - main/file.c, include/asterisk/sched.h, channels/chan_h323.c, - pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl, - channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c, - main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y, - main/db1-ast/hash/hash.c, include/asterisk/channel.h, - channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing - with cosnt pointers. This set of changes gets all of Asterisk - (minus chan_alsa for now) to compile with gcc 4.2. (closes issue - #10774, patch from qwell) - -2007-09-21 13:34 +0000 [r83400] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix video under certain circumstances. It - would have been possible for the formats on the channel to not - contain the video format. (closes issue #10782) Reported by: - cwhuang - -2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant <russell@digium.com> - - * main/asterisk.c: When daemonizing, don't change working directory - to "/". It makes it not be able to do a core dump when not - running as uid=root. (closes issue #10766, xrg) - - * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly - ask for /bin/bash, as it uses bashisms. (closes issue #10772, - reported by culrich) - -2007-09-20 17:09 +0000 [r83246] Jason Parker <jparker@digium.com> - - * apps/app_disa.c: If # is pressed after dialing an extension in - DISA, stop trying to collect more digits. (issue #10754) Reported - by: atis Patches: app_disa.c.branch.patch uploaded by atis - (license 242) app_disa.c.trunk.patch uploaded by atis (license - 242) - -2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make sure the minimum T1 timer value is - obeyed in all cases. (closes issue #10768) Reported by: flefoll - Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll - (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by - flefoll (license 244) - - * channels/chan_sip.c: Fix a minor spelling error. (closes issue - #10769) Reported by: flefoll Patches: - chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license - 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll - (license 244) - -2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant <russell@digium.com> - - * apps/app_system.c: The System() and TrySystem() applications can - take a substantial amount of time to execute while not servicing - the channel. So, put the channel in autoservice while the command - is being executed. (closes issue #10726, reported by mnicholson) - - * funcs/func_curl.c: Using curl can take a substantial amount of - time, so the channel should be autoserviced while waiting for it - to complete. (closes issue #10725, reported by mnicholson) - - * channels/chan_iax2.c: When handling a reload of chan_iax2, don't - use an ao2_callback() to POKE all peers. Instead, use an - iterator. By using an iterator, the peers container is not locked - while the POKE is being done. It can cause a deadlock if the - peers container is locked because poking a peer will try to lock - pvt structs, while there is a lot of other code that will hold a - pvt lock when trying to go lock the peers container. (reported to - me directly by Loic Didelot. Thank you for the debug info!) - - * main/manager.c: Fix up another potential race condition. Do the - loop decrementing use count on events with the eventq protected - from being changed. (reported on IRC by Ivan) - -2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Protect the CDR record from modification by - pbx_exec so that the application data contains the Queue data. - (closes issue #10761) Reported by: snar Patches: - app-queue-mixmonitor.patch uploaded by snar (license 245) - - * channels/chan_sip.c: (closes issue #10760) Reported by: dimas - Patches: chan_sip.patch uploaded by dimas (license 88) Read in - subscribecontext option in general to be the default. - -2007-09-19 09:32 +0000 [r83023-83024] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: removed comment which violates the coding - guidelines. - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h: added 'astdtmf' option to - allow configuring the asterisk dtmf detector instead of the - mISDN_dsp ones. also added the patch from irroot #10190, so that - dtmf tones detected by the asterisk detector are passed outofband - to asterisk, to make any use of dtmf tones at all. - -2007-09-19 00:19 +0000 [r82992] Russell Bryant <russell@digium.com> - - * apps/app_flash.c: Change the description of app_flash to note how - it can be a useful tool instead of just saying that it is - generally a worthless feature. (Thanks to Jim Van Meggelen for - pointing it out and providing the proposed text) - -2007-09-18 23:41 +0000 [r82961] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Initialize a variable to NULL to make the world - happy. - -2007-09-18 22:42 +0000 [r82929] Russell Bryant <russell@digium.com> - - * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle - interrupting the fgets() call when using FastAGI. This version of - the patch maintains the original behavior of the code when not - using FastAGI. (closes issue #10553) Reported by: juggie Patches: - res_agi_fgets-4.patch uploaded by juggie (license 24) - res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight - mods by me Tested by: juggie, festr - -2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey <dbailey@digium.com> - - * main/manager.c: Corrected patch applied in revision r82887. - - * main/manager.c: Fixed a bug where http manager sessions prevented - the eventq from being cleaned out because http manager sessions - do not have a valid file descriptor. - -2007-09-18 20:56 +0000 [r82867] Russell Bryant <russell@digium.com> - - * main/manager.c: Fix a memory leak that can occur on systems under - higher load. The issue is that when events are appended to the - master event queue, they use the number of active sessions as a - use count so it will know when all active sessions at the time - the event happened have consumed it. However, the handling of the - number of sessions was not properly synchronized, so the use - count was not always correct, causing an event to disappear - early, or get stuck in the event queue for forever. (closes issue - #9238, reported by bweschke, patch from Ivan, modified by me) - -2007-09-18 20:09 +0000 [r82865] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Moving the logic for handling an empty - membername to the create_member function so that there is a - common place where this occurs instead of being spread out to - several different places. - -2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: there is no need for conditional logic to - select ->interface or ->membername, snince ->membername will - always be populated - -2007-09-18 16:31 +0000 [r82802] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: When copying the contents from the wildcard - peer, do a deep copy instead of shallow copy so that it doesn't - crash when beging destroyed. (closes issue #10546, patch by me) - -2007-09-18 15:28 +0000 [r82751] Jason Parker <jparker@digium.com> - - * configs/sip.conf.sample: Correct the allowexternaldomains option - in SIP sample config. Issue 10753 - -2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in - ast_localtime() instead of a couple places in app_voicemail to - prevent the problem everywhere instead of just a couple of - places. (related to issue #10746) - - * apps/app_voicemail.c: Initialize some memory to fix crashes when - leaving voicemail. This problem was fixed by running Asterisk - under valgrind. (closes issue #10746, reported by arcivanov, - patched by me) *** IMPORTANT NOTE: We need to check to see if - this same bug exists elsewhere. - - * res/res_features.c: Handle the case where there are multiple - dynamic features with the same digit mapping, but won't always - match the activated on/by access controls. In that case, the code - needs to keep trying features for a match. (reported by Atis on - the asterisk-dev list, patched by me) - -2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: revert a change that wasn't supposed to be - committed... doh! - - * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places - where a logical member name (if specified) was not used, but - instead the direct interface was listed - -2007-09-17 02:00 +0000 [r82514] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead - of passing a NULL pointer into snprintf pass "". It makes Solaris - much happier. - -2007-09-14 21:19 +0000 [r82444] Steve Murphy <murf@digium.com> - - * main/cdr.c: closes issue #10668; thanks to arkadia for his patch; - had to leave out the bit about ending the previous cdr in the - fork; it would destroy current implementations. - -2007-09-14 21:17 +0000 [r82435] Russell Bryant <russell@digium.com> - - * configs/zapata.conf.sample: Add a note to help clarify the value - set with the echocancel option. (inspired by Malcolm's blog post - on blogs.digium.com about HPEC) - -2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Crap, I broke the build. Fixed. - - * apps/app_queue.c: Adding member name field to manager events - where they were missing before (closes issue #10721, reported by - snar) - -2007-09-14 17:48 +0000 [r82394] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: If a channel does not have an owner, do not - try to set a channel variable. This will end up making the - channel variable global, which is not right. Closes issue #10720, - patch by flefoll. - -2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant <russell@digium.com> - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add - checking for libusb here, so nobody has to deal with conflicts in - the chan_usbradio-1.4 branch every time the configure script gets - changed - - * channels/chan_usbradio.c (removed), channels/xpmr (removed), - channels/Makefile: Remove chan_usbradio from the main 1.4 branch. - It can't live here because we have a strict policy to not include - new features in release branches. However, I'm going to merge it - into trunk, and I also have a special 1.4 based branch that - includes this module. svn co - http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4 - -2007-09-14 14:42 +0000 [r82376] Mark Michelson <mmichelson@digium.com> - - * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines - (closes issue #10717, reported and patched by leedm777) - -2007-09-14 01:24 +0000 [r82368] Jim Dixon <telesistant@hotmail.com> - - * apps/app_rpt.c: Fixed problem with changes made to cdr - functionality - -2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_usbradio.c: this new driver may not live in this - branch for long (since it is a new feature), but it definitely - should not be built by default - -2007-09-14 00:34 +0000 [r82366] Jim Dixon <telesistant@hotmail.com> - - * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added), - channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added), - channels/xpmr (added), channels/xpmr/LICENSE (added), - channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample - (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added - channel driver for USB Radio device and support thereof. - -2007-09-13 23:11 +0000 [r82358] Jason Parker <jparker@digium.com> - - * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime - -2007-09-13 20:16 +0000 [r82346] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Preemptively fixing a possible segfault. It is - possible that queuename is NULL (meaning pause ALL queues), so - use q->name instead. - -2007-09-13 20:11 +0000 [r82344] Jason Parker <jparker@digium.com> - - * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when - mutliple threads tried to close the same file. Do we actually - need the locking here? What happens if you open the same file - twice, and two threads try to write to it at the same time? Is - fputs() going to write out the entire line at once? I suspect - that it could be possible for the second fopen to run during the - first fputs, so the position could be in the middle of the - previously written line... Issue 10347, initial patch by - explidous (but I removed all of the paranoia stuff..) - -2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant <russell@digium.com> - - * main/astobj2.c: resolve a warning when not building under dev - mode - - * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile - in tracking astobj2 statistics if dev-mode is enabled. Also, when - dev mode is enabled, register the CLI command that can be used to - run the astobj2 test and print out statistics. - -2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming <kpfleming@digium.com> - - * /, LICENSE: Merged revisions 82334 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007) - | 2 lines clarify the OpenSSL and OpenH323 license exceptions - ........ - -2007-09-13 16:25 +0000 [r82326] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Added logic to handle the unlikely case that - someone has two queues with the same name. Asterisk will log a - warning message letting the user know that one was already - defined with that name and is it skipping all further instances. - This also will work for realtime queues but in order for that to - happen, the user would have to trigger a perfectly timed reload - as a realtime queue is being looked up, which is highly unlikely - (but taken care of nonetheless). - -2007-09-13 11:47 +0000 [r82309] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Closes issue #9401, reported and patched - by irrot, with slight modifications by me. Handle DTMF sent by - Asterisk properly. - -2007-09-12 21:56 +0000 [r82296] Russell Bryant <russell@digium.com> - - * res/res_agi.c: Fix a check of the wrong pointer, as pointed out - by an XXX comment left in the code. The problem was harmless, - however. - -2007-09-12 21:28 +0000 [r82291] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone - files - -2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard <dhubbard@digium.com> - - * apps/app_meetme.c: remove a race condition for the creation of - recordthread's, and fix a small memory leak. This closes issue# - 10636 - -2007-09-12 20:12 +0000 [r82285] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/private.h, main/stdtime/tzfile.h, - include/asterisk/localtime.h, main/stdtime/localtime.c: Working - on issue #10531 exposed a rather nasty 64-bit issue on - ast_mktime, so we updated the localtime.c file from source. Next - we'll have to write ast_strptime to match. - -2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Clean up the output of "asterisk -h". This - tweaks the wording and wraps lines at 80 characters. (closes - issue #10699, seanbright) - - * res/res_agi.c: revert patch from issue #10553, as someone not - using fastagi reported that this broke their system. - -2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Accidentally committed changes to - app_voicemail which do NOT need to be in the 1.4 branch yet. - reverting... - - * apps/app_voicemail.c, apps/app_queue.c: We should only initialize - a realtime queue when it is allocated, not every time we access - it. This prevents the members ao2_container from being - reallocated every time the queue is accessed. I also removed a - debug message I had accidentally left in on a previous commit. - -2007-09-11 22:37 +0000 [r82267] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of - these calls was reading bogus memory ... - -2007-09-11 21:41 +0000 [r82265] Joshua Colp <jcolp@digium.com> - - * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue - #10679) Reported by: andrew Build under dev mode when K6OPTS is - enabled. - -2007-09-11 20:49 +0000 [r82263] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix another missing unref of member objects. - This one was pointed out by Marta. When building the outgoing - list in try_calling(), a member reference is stored in each - outgoing entry. However, when this list got destroyed, the - reference was not released. - -2007-09-11 20:36 +0000 [r82261] Steve Murphy <murf@digium.com> - - * main/cdr.c: this change should fix issue # 10659 -- what I worry - about is how many other bug reports it may generate. Hopefully, - we can please the/a majority. Hopefully. We shall see. Calls not - marked ANSWERED and with only one channel name will not be - posted. This should eliminate the double CDR's. - -2007-09-11 16:05 +0000 [r82252] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: All instances of ao2_iterators which were just - named 'i' have been renamed to 'mem_iter' so that when refcounted - queues are merged into trunk, there will be little confusion - regarding iterator names, especially when a queue and member - iterator are used in the same function. - -2007-09-11 16:03 +0000 [r82250] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: The sample dundi.conf claims support for a - wildcard peer entry - [*], but the code did not support it. This - patch makes it work. (closes issue #10546, patch by dds, with - some changes by me) - -2007-09-11 16:01 +0000 [r82249] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a - hold/retrieve issue. - -2007-09-11 15:26 +0000 [r82245] Russell Bryant <russell@digium.com> - - * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches: - res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by: - juggie When using fastagi, fgets() can return before a full line - is read. Add explicit handling for the case where it gets - interrupted. - -2007-09-11 14:56 +0000 [r82243] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich - Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich - (license 176) Don't leak memory when unloading DUNDi. - -2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Add a couple more missing unrefs of queue - member objects - - * apps/app_queue.c: Add a missing unref of a queue member in an - error handling block - - * apps/app_queue.c: Document why membercount can not simply be - replaced by ao2_container_count() - - * main/astobj2.c: backport astobj2 race condition fix. This - function is the exact same as trunk so it applies here as well. - -2007-09-10 18:02 +0000 [r82155] Tilghman Lesher <tlesher@digium.com> - - * apps/app_queue.c: Convert struct member to use refcounts (closes - issue #10199) - -2007-09-10 15:02 +0000 [r82091] Mark Michelson <mmichelson@digium.com> - - * configs/misdn.conf.sample: Removing non-existent options from - misdn configuration sample. (closes issue #10678, reported and - patched by IgorG) - -2007-09-09 02:35 +0000 [r82028] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/lock.h: Fix inline compiles on really old - compilers (who uses gcc 2.7 anymore, really?) - -2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit() - did not free the entry it removed. - - * .cleancount: (closes issue #10672) Bump the cleancount so that a - "make clean" will be forced. This is needed because my fix in - revision 81599 made a change to a data structure in file.h, and - since file dependency tracking is only on with dev-mode enabled, - file format modules that don't get rebuilt may crash, as is the - case with this issue. This makes me wonder - how much faster does - the code build without the file dependency tracking enabled? If - it doesn't make much of a difference, then it may be worth just - keeping it on all of the time, or perhaps just not in release - tarballs, so that this type of issue is avoided. - -2007-09-07 19:48 +0000 [r81923] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used - as the mixmonitor filename. This moves the setting of the - MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch - by sim. - -2007-09-07 15:25 +0000 [r81886] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample: Moving the explanation for joinempty - to a more appropriate place - -2007-09-06 22:28 +0000 [r81832] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: (closes issue #9724, closes issue #10374) - Reported by: kenw Patches: 9724.txt uploaded by russell (license - 2) Tested by: kenw, russell Resolve a deadlock that occurs when - doing a SIP transfer to parking. I come across this type of - deadlock fairly often it seems. It is very important to mind the - boundary between the channel driver and the core in respect to - the channel lock and the channel-pvt lock. Channel drivers lock - to lock the pvt and then the channel once it calls into the core, - while the core will do it in the opposite order. The way this is - avoided is by having channel drivers either release their pvt - lock while calling into the core, or such as in this case, - unlocking the pvt just long enough to acquire the channel lock. - -2007-09-06 22:05 +0000 [r81778-81826] Jason Parker <jparker@digium.com> - - * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS - for ASTLDFLAGS. This adds LDOPTS - - * include/asterisk/astobj2.h: This should fix a build issue that - people building against uClibc were seeing with the addition of - astobj2 - -2007-09-06 19:40 +0000 [r81776] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: (closes issue #10122) Reported by: - stevefeinstein Patches: meetme-unmute-manager.diff uploaded by - qwell (license 4) Tested by: stevefeinstein After looking over - the code I agree with Qwell. Setting the file descriptor to - conference each time just causes a fight back and forth. - -2007-09-06 16:56 +0000 [r81743] Philippe Sultan <philippe.sultan@gmail.com> - - * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string - length fixes. Removed an unused variable in aji_client structure - (context) - -2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixes an issue where valid DTMF had to be - pressed twice to exit a queue if a member's phone was ringing. - (closes issue #10655, reported by strider2k, patched by me) - - * res/res_features.c: Fixes a memory leak (closes issue #10658, - reported and patched by Ivan) - -2007-09-06 14:20 +0000 [r81650] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 - - and Google's XMPP server complaint, if set, the 'from' attribute - must be set to the user's full JID. - -2007-09-05 20:53 +0000 [r81599] Russell Bryant <russell@digium.com> - - * include/asterisk/file.h, main/say.c, res/res_features.c, - main/file.c, include/asterisk/channel.h: Fix an issue that can - occur when you do an attended transfer to parking. If you - complete the transfer before the announcement of the parking spot - finishes, then the channel being parked will hear the remainder - of the announcement. These changes make it so that will not - happen anymore. Basically, res_features sets a flag on the - channel is playing the announcement to so that the file streaming - core knows that it needs to watch out for a channel masquerade, - and if it occurs, to abort the announcement. (closes BE-182) - -2007-09-05 17:18 +0000 [r81569] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/lock.h: Solaris x86 compatibility fix - -2007-09-05 15:19 +0000 [r81525] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixing the build... - -2007-09-05 15:14 +0000 [r81523] Jason Parker <jparker@digium.com> - - * channels/chan_phone.c: Do not try to unregister a NULL channel - tech. Also changed load_module function to use defines rather - than numbers for return values. Issue 10651, patch by - rbraun_proformatique, with additions by me. - -2007-09-05 15:03 +0000 [r81520] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to - only count members who are logged in and available. (related to - issue #10652, reported by wuwu) - -2007-09-05 13:11 +0000 [r81492] Joshua Colp <jcolp@digium.com> - - * main/channel.c: (closes issue #10650) Reported by: tacvbo Only - print out that the spy was removed while holding the spy lock. - -2007-09-04 20:54 +0000 [r81453-81455] Jason Parker <jparker@digium.com> - - * apps/app_followme.c: Rather than attempt to play a file, we can - just check whether it exists. Issue 10634, patch by me, testing - by pabelanger, sanity checked by bweschke - - * configs/followme.conf.sample: Change default followme config file - to point to the correct files. Issue 10644, patch by pabelanger - -2007-09-04 18:37 +0000 [r81448] Russell Bryant <russell@digium.com> - - * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c: - Remove the typedefs on ao2_container and ao2_iterator. This is - simply because we don't typedef objects anywhere else in - Asterisk, so we might as well make this follow the same - convention. - -2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: there is no point in sending 401 - Unauthorized to a UAS that sent us a properly-formatted - Authentication header with the expected username and nonce but an - incorrect response (which indicates the shared secret does not - match)... instead, let's send 403 Forbidden so that the UAS - doesn't retry with the same authentication credentials repeatedly - -2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: (closes issue #10632) Reported by: - jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded - by jamesgolovich (license 176) Fix memory leak when unloading - chan_iax2. The firmware files were not being freed. - - * main/channel.c: (closes issue #10476) Reported by: mdu113 Only - look for the end of a digit when waiting for a digit. This in - turn disables emulation in the core. - - * main/dns.c: (closes issue #10610) Reported by: john Patches: - dns.c.patch uploaded by john (license 218) Tested by: mvanbaak - Don't return a match if no SRV record actually exists. - -2007-09-03 18:57 +0000 [r81433] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Remove a couple of calls to - ast_string_field_free_pools() on peers in error handling blocks - in the code for building peers. The peer object destructor does - this and doing it twice will cause a crash. (closes issue #10625, - reported by and patched by pnlarsson) - -2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Changed a comment to be more accurate. (really - this is just a test to make sure I can commit properly from home) - - * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr - into ao2_match_by_addr and making it available everywhere since - it could be a handy callback to have - -2007-08-31 21:27 +0000 [r81418] Russell Bryant <russell@digium.com> - - * include/asterisk/astobj2.h: Remove references to a debugging - parameter that does not exist - -2007-08-31 19:48 +0000 [r81416] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixed broken behavior of a reload on realtime - queues. Prior to this patch, if a reload was issued and a - realtime queue had callers waiting in it, then the queue would be - removed from the queue list, but it would not actually be freed - (in fact, a debug message warning about a memory leak would come - up). With this patch, reloads do not touch realtime queues at - all. - -2007-08-31 19:16 +0000 [r81415] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_logic.c: The IF() function was not allowing true - values that had embedded colons (closes issue #10613) - -2007-08-31 18:44 +0000 [r81412] Jason Parker <jparker@digium.com> - - * apps/app_dial.c: Re-order dial options to be in line with the - existing alpha order. Issue 10621, initial patch by junky - -2007-08-31 17:38 +0000 [r81410] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command - available. Closes issue 10548, reported by keepitcool. - -2007-08-31 15:53 +0000 [r81406] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c: Make it the engine's responsible to check for - the presence of results. - -2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: add missing "transcoder show" (and deprecated - "show transcoder") CLI commands that were in 1.2 but never added - to 1.4 - -2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (closes issue #10618) Reported by: dimas - Don't pass through the stopped sounds frame.... just drop it. - - * res/res_features.c: (closes issue #10009) Reported by: dimas - Don't output a bridge failed warning message if it failed because - one of the channels was part of the masquerade process. That is - perfectly normal. - -2007-08-30 22:05 +0000 [r81397] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Removing an extraneous (and possibly - misleading) log message. Firstly, if the announce file isn't - found, the streaming functions will report it. Secondly, not all - non-zero returns from play_file mean that the announce file - wasn't found. Positive return values simply mean that a digit was - pressed (most likely to skip through the announcement). (closes - issue #10612, reported and patched by dimas) - -2007-08-30 21:23 +0000 [r81395] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: (closes issue #10514) Reported by: casper - Patches: chan_sip.c.80129.diff uploaded by casper (license 55) - Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible - for it to ever be that value. - -2007-08-30 21:11 +0000 [r81392] Steve Murphy <murf@digium.com> - - * main/cdr.c: via issue 10599, where 'CDR already initialized' - messages are being generated. Since all channels will have an - init'd CDR attached at creation time, this message is now - particularly useless. Removed. - -2007-08-30 15:38 +0000 [r81383] Russell Bryant <russell@digium.com> - - * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING - define. (closes issue #10559, paravoid) - -2007-08-30 15:35 +0000 [r81381] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Changed some manager event messages to reflect - whether a queue member is a realtime member or not - -2007-08-30 15:33 +0000 [r81379] Russell Bryant <russell@digium.com> - - * configs/modem.conf.sample (removed), configs/enum.conf.sample, - configs/extensions.ael.sample: Fix a typo, update a reload - command, and remove an unused configuration file. (closes issue - #10606, casper) - -2007-08-30 14:53 +0000 [r81375] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: (closes issue #10603) Reported by: jmls Patches: - pbx.diff uploaded by jmls (license 141) Backport changes from - 81372. Add REASON dialplan variable for when an originated call - fails and the failed extension is executed. - -2007-08-30 14:43 +0000 [r81373] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: Fixed some warnings. - -2007-08-30 14:23 +0000 [r81369] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (issue #10599) Reported by: dimas Handle the - -1 control subclass during feature dialing (it indicates to stop - sounds). - -2007-08-30 08:31 +0000 [r81367] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe - issue where a misdn_read would lock the channel, but read would - not return because it blocks. later chan_misdn would try to queue - a frame like a AST_CONTROL_ANSWER which could result in a - deadlock situation. misdn_read will now not block forever - anymore, and we don't queue the ANSWER frame at all when we - already was called with misdn_answer -> answer would be called - twice. Also we don't explicitly send a RELEASE_COMPLETE on - receiption of a RELEASE anymore, because mISDN does that for us, - this resulted in a problem on some switches, which would block - our port after some calls for a short while. - -2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: This patch, in essence, will correctly pause a - realtime queue member and reflect those changes in the realtime - engine. (issue #10424, reported by irroot, patch by me) This - patch creates a new function called update_realtime_member_field, - which is a generic function which will allow any one field of a - realtime queue member to be updated. This patch only uses this - function to update the paused status of a queue member, but it - lays the foundation for persisting the state of a realtime member - the same way that static members' state is maintained when using - the persistentmembers setting - - * apps/app_queue.c: Changed some tabs to spaces - -2007-08-29 15:57 +0000 [r81342] Russell Bryant <russell@digium.com> - - * main/Makefile: If chan_h323 is not being built, don't use g++ to - do the final link of Asterisk. (in response to a question on the - asterisk-dev list) - -2007-08-29 15:52 +0000 [r81340] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: This fix creates a more accurate way of - detecting whether realtime members were deleted. (closes issue - 10541, reported by Alric, patched by me) The REALLY nice things - about this patch is that queue members now have a "realtime" - field which will be true if the member is a realtime member. This - means we can check this value prior to certain processing if it - should ONLY be done for realtime members. - -2007-08-29 14:13 +0000 [r81331] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make - rtp timeouts work even if two RTP streams are directly bridged in - the RTP stack. - -2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Change the message about receiving a - mini-frame before the first full voice frame to a DEBUG message. - - * pbx/pbx_dundi.c: revert unintentional changes in rev 81226 - - * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian - tones. (closes issue #7953, hanabana) - -2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson <mmichelson@digium.com> - - * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using - res_config_mysql (closes issue #10573, reported by chrisvaughan, - patch suggested by chrisvaughan as well) - - * apps/app_queue.c: Resolve a potential deadlock. In this case, a - single queue is locked, then the queue list. In changethread(), - the queue list is locked, and then each individual queue is - locked. Under the right circumstances, this could deadlock. As - such, I have unlocked the individual queue before locking the - queue list, and then locked the queue back after the queue list - is unlocked. - - * channels/chan_agent.c: DTMF begin frames should be ignored so - that when an agent acks a call with the '#' key, he doesn't cause - a queue's announce file to be interrupted. Also went ahead and - did the same for the '*' key and for ending a call. (closes issue - #10528, reported by deskhack, patched by me) - -2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most - of the memory it allocates on unload and should be fixed (when - I'm not in the middle of something else ...). - - * pbx/pbx_dundi.c: explicity define a variable as a boolean - - * res/res_musiconhold.c: (closes issue #10419) Reported by: - mustardman Patches: asterisk-mohposition.diff.txt uploaded by - jamesgolovich (license 176) This patch fixes a few problems with - music on hold. * Fix issues with starting at the beginning of a - file when it shouldn't. * Fix the inuse counter to be decremented - even if the class had not been set to be deleted when not in use - anymore * Don't arbitrarily limit the number of MOH files to 255 - -2007-08-27 15:01 +0000 [r81012] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang - Patches: chan_sip-ChannelReload-20080825.patch uploaded by - jesselang (license 202) Remove an extra \r\n to make the - ChannelReload event conform with every other event. - -2007-08-27 14:55 +0000 [r81010] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Found a case where the queue's membercount is - off. It does not take into account dynamic members on a reload. - -2007-08-27 13:20 +0000 [r80974] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct - jitter value output in the CLI to be as expected. - -2007-08-26 18:11 +0000 [r80932] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Remove an extra signal_condition() for the - scheduler thread. (closes issue #10564, patch from casper) - -2007-08-25 17:37 +0000 [r80895] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix some issues with the handling of the - scheduler in chan_iax2. Most of the places that scheduled items - to be executed by the scheduler thread did not signal the - scheduler thread to wake up so that it could recalculate the time - until the next action. These changes will make the scheduler - thread more responsive and ensure that actions get executed as - close to when intended as possible instead of it being possible - for very long delays. - -2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard <dhubbard@digium.com> - - * apps/app_zapateller.c: An empty string is an empty callerid ... - so zap it. This closes issue #10502, which was pointed out by - dswartz. Thank you, and may the swartz be with you - -2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are - available when Asterisk first starts, then don't give up on - poking peers. Allow the poke to get rescheduled so that it will - work once the dnsmgr is able to resolve the host. (closes issue - #10521, patch by jamesgolovich) - - * main/dsp.c: Improve the debouncing logic in the DTMF detector to - fix some reliability issues. Previously, this code used a shift - register of hits and non-hits. However, if the start of the digit - isn't clean, it is possible for the leading edge detector to miss - the digit. These changes replace the flawed shift register logic - and also does the debouncing on the trailing edge as well. - (closes issue #10535, many thanks to softins for the patch) - -2007-08-24 19:52 +0000 [r80818] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: A minor correction to the available logic of - autofill. If a queue member is paused, they're not really - "available" so don't count them as such. Somewhat related to - issue #10155 - -2007-08-24 18:52 +0000 [r80789] Steve Murphy <murf@digium.com> - - * main/cdr.c: From a complaint by jmls, I realize that the message - in cdr_disposition is unnecessary. To get failure disposition, - just return -1; no use having more than one case do that. - -2007-08-24 15:51 +0000 [r80750] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail. - -2007-08-24 15:41 +0000 [r80747] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with - the code, instead of only in documentation (closes issue #10549) - -2007-08-24 15:28 +0000 [r80722] Russell Bryant <russell@digium.com> - - * utils/ael_main.c: Tweak the formatting of this MODULEINFO block. - I think this would have caused a "*" to get in the - menuselect-tree file. - -2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy <murf@digium.com> - - * utils/ael_main.c: This change addresses JerJer's complaint that - aelparse builds and installs even if pbx_ael is unchecked in the - menuselect stuff. - - * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6: - backport of 80649, a fix to an unreported problem in the ael - parser, that results in a crash on a 64bit machine - -2007-08-24 11:42 +0000 [r80661] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are - answered too early, which results in CDRs wrongly stating that a - call was ANSWERED when the calling party cancelled a call before - before being established. We must not answer the call upon - reception of a 'transport-accept' iq packet, but this packet - still needs to be acknowledged, otherwise the remote peer would - close the call (like in #8970). - -2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without - warnings - - * channels/chan_misdn.c: make chan_misdn compile without warnings - -2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant <russell@digium.com> - - * include/asterisk/features.h, res/res_features.c: When executing a - dynamic feature, don't look it up a second time by digit pattern - after we already looked it up by name. This causes broken - behavior if there is more than one feature defined with the same - digit pattern. (closes issue #10539, reported by bungalow, patch - by me) - - * funcs/func_timeout.c: Revert very broken fix for issue #10540 ... - none of these values take ms so I don't know what I was thinking - - * funcs/func_timeout.c: Fix func_timeout to take values in floating - point so 1.5 actually means 1.5 seconds instead of being rounded. - (closes issue #10540, reported by spendergrass, patch by me) - -2007-08-23 17:14 +0000 [r80505-80507] Jason Parker <jparker@digium.com> - - * /: *sigh* - - * /: use autotagged externals - -2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: report the actual channel number that was - unregistered, instead of assuming that the interface list - consists of channels 1 through <x> with no gaps in the sequence - -2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix some code where it was possible for a - reference to a peer to not get released when it should. Thank you - to Marta Carbone for pointing this out! - - * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c: - This is a hack to maintain old behavior of chan_iax2. This - ensures that if the peers and users are being stored in a linked - list, that they go in the list in the same order that the older - code used. This is necessary to maintain the behavior of which - peers and users get matched when traversing the container. - - * res/res_agi.c: Revert res_agi fix that didn't quite work until we - get it right ... - - * include/asterisk/astobj2.h: Add some more documentation on - iterating ao2 containers. The documentation implies that is - possible to miss an object or see an object twice while - iterating. After looking through the code and talking with - mmichelson, I have documented the exact conditions under which - this can happen (which are rare and harmless in most cases). - - * main/astobj2.c: When converting this code to use the list macros, - I changed it so objects are added to the head of a bucket instead - of the tail. However, while looking over code with mmichelson, we - noticed that the algorithm used in ao2_iterator_next requires - that items are added to the tail. This wouldn't have caused any - huge problem, but it wasn't correct. It meant that if an object - was added to a container while you were iterating it, and it was - added to the same bucket that the current element is in, then the - new object would be returned by ao2_iterator_next, and any other - objects in the bucket would be bypassed in the traversal. - - * channels/chan_sip.c: Don't crash when using realtime in chan_sip - without an insecure setting in the database. (closes issue - #10348, reported by link55, fixed by me) - - * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h - (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h, - channels/chan_iax2.c: Merge changes from - team/russell/iax_refcount. This set of changes fixes problems - with the handling of iax2_user and iax2_peer objects. It was very - possible for a thread to still hold a reference to one of these - objects while a reload operation tries to delete them. The fix - here is to ensure that all references to these objects are - tracked so that they can't go away while still in use. To - accomplish this, I used the astobj2 reference counted object - model. This code has been in one of Luigi Rizzo's branches for a - long time and was primarily developed by one of his students, - Marta Carbone. I wanted to go ahead and bring this in to 1.4 - because there are other problems similar to the ones fixed by - these changes, so we might as well go ahead and use the new - astobj if we're going to go through all of the work necessary to - fix the problems. As a nice side benefit of these changes, peer - and user handling got more efficient. Using astobj2 lets us not - hold the container lock for peers or users nearly as long while - iterating. Also, by changing a define at the top of chan_iax2.c, - the objects will be distributed in a hash table, drastically - increasing lookup speed in these containers, which will have a - very big impact on systems that have a large number of users or - peers. The use of the hash table will be made the default in - trunk. It is not the default in 1.4 because it changes the - behavior slightly. Previously, since peers and users were stored - in memory in the same order they were specified in the - configuration file, you could influence peer and user matching - order based on the order they are specified in the configuration. - The hash table does not guarantee any order in the container, so - this behavior will be going away. It just means that you have to - be a little more careful ensuring that peers and users are - matched explicitly and not forcing chan_iax2 to have to guess - which user is the right one based on secret, host, and access - list settings, instead of simply using the username. If you have - any questions, feel free to ask on the asterisk-dev list. - - * res/res_agi.c: Juggie in #asterisk-dev was reporting problems - where fgets would return without reading the whole line when - using fastagi. When this happens, errno was set to EINTR or - EAGAIN. This patch accounts for the possibility and lets fgets - continue in that case. - -2007-08-22 18:53 +0000 [r80302-80330] Jason Parker <jparker@digium.com> - - * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h, - build_tools/strip_nonapi, build_tools/prep_moduledeps, - build_tools/make_buildopts_h: Fix a few build issues in Solaris - (and likely others). Use GREP and ID variables from autoconf. - Reported to me in #asterisk-dev I forgot who reported this - - sorry. :( - - * Makefile: Change a syntax that the GNU make in Solaris dislikes. - - * build_tools/make_version: Fix a bashism (we explicitly request - /bin/sh). Remove some oddly placed quotes I found in passing. - -2007-08-22 16:21 +0000 [r80257] Russell Bryant <russell@digium.com> - - * Makefile: Honor the contents of the COPTS variable as custom - target CFLAGS. Apparently this is what openwrt does. (reported by - Brian Capouch on the asterisk-dev list, patch by me) - -2007-08-22 16:14 +0000 [r80255] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: (closes issue #10526) Reported by: sinistermidget - Revert commit from issue #10355 and return timestamp skew to 640. - -2007-08-21 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.11 released. - -2007-08-21 18:42 +0000 [r80183] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Don't record SIP dialog history if it's not - turned on. Also, put an upper limit on how many history entires - will be stored for each SIP dialog. It is currently set to 50, - but can be increased if deemed necessary. (closes issue #10421, - closes issue #10418, patches suggested by jmoldenhauer, patches - updated by me) (Security implications documented in AST-2007-020) - -2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy <murf@digium.com> - - * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing - the diffs from ulaw.h and alaw.h for now; accidentally added them - in 80166 - - * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h: - This patch solves problem 1 in 8126; it should not slow down the - alaw codec, but should prevent signal degradation via multiple - trips thru the codec. Fossil estimates the twice thru this codec - will prevent fax from working. 4-6 times thru would result - hearable, noticeable, voice degradation. - -2007-08-21 15:22 +0000 [r80132] Russell Bryant <russell@digium.com> - - * channels/chan_mgcp.c: Don't try to dereference the owner channel - when it may not exist (issue #10507, maxper) - -2007-08-21 15:03 +0000 [r80130] Jason Parker <jparker@digium.com> - - * configs/cdr.conf.sample: (issue #10510) Reported by: casper - Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few - errors in sample cdr config file. - -2007-08-20 21:57 +0000 [r80088] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix the build of app_queue - -2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: After a discussion on #asterisk-dev, it was - decided that this should be in 1.4 as well. (issue #10424, - reported and patched by irroot) - - * apps/app_queue.c: Found a pointless ternary if. member->dynamic - was set to 1 and has no opportunity to change between then and - this line, so "dynamic" will ALWAYS be output. - -2007-08-20 16:08 +0000 [r80047] Jason Parker <jparker@digium.com> - - * configs/extensions.conf.sample: (issue #10499) Reported by: - casper Patches: extensions.conf.sample.diff uploaded by casper - (license 55) Update CLI examples in extensions.conf.sample to - reflect command changes. - -2007-08-20 15:34 +0000 [r80044] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Ukrainian language voicemail support. - (closes issue #10458, reported and patched by Oleh) - -2007-08-20 02:42 +0000 [r79998] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by - snuffy via IRC) - -2007-08-18 14:30 +0000 [r79947] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Don't allocate vmu for messagecount when we - could just use the stack instead (closes issue #10490) Also, - remove a useless (and leaky) SQLAllocHandle (closes issue #10480) - -2007-08-17 21:01 +0000 [r79912] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Avoid a crash in the handling of DTMF based - Caller ID. It is valid for ast_read to return NULL in the case - that the channel has been hung up. (crash reported by - anonymouz666 on IRC in #asterisk-dev) - -2007-08-17 19:14 +0000 [r79906] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Patch allows for more seamless transition - from file storage voicemail to ODBC storage voicemail. If a - retrieval of a greeting from the database fails, but the file is - found on the file system, then we go ahead an insert the greeting - into the database. The result of this is that people who switch - from file storage to ODBC storage do not need to rerecord their - voicemail greetings. - -2007-08-17 19:12 +0000 [r79902-79904] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h: - Don't send a semicolon over the wire in sip notify messages. - Caused by fix for issue 9938. I basically took the code that - existed before 9938 was fixed, and copied it into a new function - - ast_unescape_semicolon There should be very few places this - will be needed (pbx_config does NOT need this (see issue 9938 for - details)) Issue 10430, patch by me, with help/ideas from murf - (thanks murf). - - * channels/chan_local.c: Re-add the setting of callerid name and - number. Issue 10485, reported by and fix explained by paradise. - -2007-08-17 13:37 +0000 [r79857] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix some crashes in chan_sip. This patch - changes various places that add items to the scheduler to ensure - that they don't overwrite the ID of a previously scheduled item. - If there is one, it should be removed. (closes issue #10391, - closes issue #10256, probably others, patch by me) - -2007-08-17 08:22 +0000 [r79833] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: sometimes we don't need to signal dtmf - tones to asterisk, we just want them to go through as inband. - Otherwise they might be generated by the other channel partner - and then there is a double tone. - -2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Fix a little race condition that could - cause a crash if two channels had MOH stopped at the same time - that were using a class that had been marked for deletion when - its use count hits zero. - - * res/res_musiconhold.c: This patch fixes a bug where reloading the - module with "module reload" did not delete classes from memory - that were no longer in the config. This patch fixes that problem - as well as another one. Previously, if you reloaded MOH using the - "moh reload" CLI command, which behaved differently than "module - reload ...", MOH had to be stopped on every channel and started - again immediately. However, there was no way to tell what class - was being used, so they would all fall back to the default class. - (closes issue #10139) Reported by: blitzrage Patches: - asterisk-10139-advanced.diff.txt uploaded by jamesgolovich - (license 176) Tested by: jamesgolovich - - * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were - introduced by making frame handling and scheduling - multi-threaded. Unfortunately, we have to do some expensive - deadlock avoidance when queueing frames on to the ast_channel - owner of the IAX2 pvt struct. This was already handled for - regular frames, but ast_queue_hangup and ast_queue_control were - still used directly. Making these changes introduced even more - places where the IAX2 pvt struct can disappear in the context of - a function holding its lock due to calling a function that has to - unlock/lock it to avoid deadlocks. I went through and fixed all - of these places to account for this possibility. (issue #10362, - patch by me) - -2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Fixes a problem where agents would get - stuck busy due to their wrapuptime being longer than the queue's - wrapuptime and ringinuse=no for the queue. (closes issue #10215, - reported by Doug, repaired by me) Special thanks to fkasumovic - for pointing out the source of the problem and to bweschke for - helping to come up with a solution! - - * apps/app_voicemail.c: base_encode is not trying to open a log - file, so we should not call it a log file in the warning. - (related to issue #10452, reported by bcnit) - -2007-08-16 09:37 +0000 [r79665] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: A fix for two critical problems detected while - working with Daniel McKeehan in issue #10184. Upon priority - change, the resource list is not NULL terminated when moving an - item to the end of the list. This makes Asterisk endlessy loop - whenever it needs to read the list. Jids with different resource - and priority values, like in Gmail's and GoogleTalk's jabber - clients put that problem in evidence. Upon reception of a 'from' - attribute with an empty resource string, Asterisk crashes when - trying to access the found->cap pointer if the resource list for - the given buddy is not empty. This situation is perfectly valid - and must be handled. The Gizmoproject's jabber client put that - problem in evidence. Also added a few comments in the code as - well as a handle for the capabilities from Gmail's jabber client, - which are stored in a caps:c tag rather than the usual c tag. - Closes issue #10184. - -2007-08-16 08:21 +0000 [r79642] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when - we want to send IA5 Chars. - -2007-08-15 14:40 +0000 [r79553] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: (closes issue #10440) Reported by: irroot (closes - issue #10454) Reported by: flo_turc Increase maximum timestamp - skew to 120. 20 was apparently far too low. - -2007-08-15 14:26 +0000 [r79527] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixed an error in the Russian language - voicemail intro. (issue #10458, reported and patched by Oleh) - -2007-08-15 14:18 +0000 [r79523] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: (closes issue #10456) Reported by: irroot - Patches: sip_timeout.patch uploaded by irroot (license 52) Change - hardcoded timer value to defined value. I'm doing this in 1.4 as - well so if it needs to be changed in the future this place would - not have been forgotten. - -2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix another spot where an iax2_peer would - be leaked if realtime was in use. - - * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2 - related to the use of realtime. I found these while working on - iax2_peer object reference tracking. - -2007-08-14 15:27 +0000 [r79397] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (closes issue #10415) Reported by: atis - Revert fix for #10327 as it causes more issues then it solves. - -2007-08-13 22:40 +0000 [r79363] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: memset really, really needs to be used here. - -2007-08-13 21:57 +0000 [r79334] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c, apps/app_speech_utils.c, - include/asterisk/speech.h: Instead of accepting a single DTMF - character accept a full string. - -2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Don't call find_peer in - registry_authrequest with the pvt lock held to avoid a deadlock. - - * channels/chan_iax2.c: Release the pvt lock before calling - find_peer in register_verify to avoid a deadlock. Also, remove - some unnecessary locking in auth_fail that was only done - recursively. - - * channels/chan_iax2.c: Don't call find_peer within update_registry - with a pvt lock held. This can cause a deadlock as the code will - eventually call find_callno. - - * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I - have tracked them down to a single core issue. You can not call - find_callno() while holding a pvt lock as this function has to - lock another (every) other pvt lock. Doing so can lead to a - classic deadlock. So, I am tracking down all of the code paths - where this can happen and fixing them. The fix I committed - earlier today was along the same theme. This patch fixes some - code down the path of authenticate_reply. - -2007-08-13 17:49 +0000 [r79255] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest21 (added), - pbx/ael/ael-test/ref.ael-test19, - pbx/ael/ael-test/ael-vtest21/extensions.ael (added), - pbx/ael/ael-test/ael-vtest21 (added), - pbx/ael/ael-test/ref.ael-vtest17, - pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c, - pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch - fixes bug 10411. I added a new regression test, some regression - test cleanups - -2007-08-13 15:28 +0000 [r79214] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a potential deadlock in socket_process. - check_provisioning can eventually call find_callno. You can't - hold a pvt lock while calling find_callno because it goes through - and locks every single one looking for a match. - -2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c, apps/app_speech_utils.c, - include/asterisk/speech.h: Add an API call to allow the engine to - know that DTMF was received. - - * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c, - channels/chan_local.c, channels/chan_misdn.c, - channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_h323.c, channels/chan_gtalk.c, - channels/chan_iax2.c: (closes issue #10437) Reported by: haklin - Don't set the callerid name and number a second time on a newly - created channel. ast_channel_alloc itself already sets it and - setting it twice would cause a memory leak. - -2007-08-11 05:23 +0000 [r79142] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Ensure the connection gets marked as used at - allocation time (closes issue #10429, report and fix by - mnicholson) - -2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy <murf@digium.com> - - * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From - a user complaint on #asterisk, I have forced pbx_spool to explain - what reason codes mean, when they are logged - - * main/cdr.c: Re bug behavior mentioned in #asterisk, made this - tweak to code, to prevent hundreds of log messages from being - generated - - * main/cdr.c: This will help debug; from a question asked on - #asterisk - -2007-08-10 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.10.1 released. - -2007-08-10 15:20 +0000 [r78995] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: The last set of changes that I made to - "core show locks" made it not able to track mutexes unless they - were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized - with ast_mutex_init() were not tracked. It should work now. - -2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp <jcolp@digium.com> - - * main/file.c: Don't bother having the core pass through or emulate - begin DTMF frames when in an ast_waitstream. It only cares about - the end of DTMF. - - * configs/queues.conf.sample: (closes issue #10422) Reported by: - bhowell Add note to sample configuration about module load order - and how it can cause perfectly good queue members to be marked as - invalid. - -2007-08-10 13:24 +0000 [r78936] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser - information element. We send them now also in the disconnect - message. - -2007-08-09 23:47 +0000 [r78907] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Improved a bit of logic regarding - comma-separated mailboxes in has_voicemail. Also added some - braces to some compound if statements since unbraced if - statements scare me in general. - -2007-08-09 23:10 +0000 [r78891] Steve Murphy <murf@digium.com> - - * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty - output - -2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removing some extra debug code I left in my - last commit - - * apps/app_voicemail.c: Quite a few changes regarding IMAP storage. - 1. instead of using inboxcount as the core message counting - function, we use messagecount instead. This makes it possible to - count messages in folders besides just INBOX and Old. 2. - inboxcount and hasvoicemail now use messagecount as their means - of determining return values. 3. Added a copy_message function - for IMAP storage. Unfortunately I don't have the means to test - it, but it seems like a pretty straightforward function. 4. - Removed a #ifndef IMAP_STORAGE and matching #endif from - leave_voicemail for a couple of reasons. One, we want to support - copying mail to multiple IMAP boxes, and two, IMAP was broken - because a STORE macro had been moved into this section of code. - - * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I - put some new code in the wrong place and so I've reverted the - canreinvite section to how it was and put my new code where it - should be. - -2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: add a comment to indicate that inboxcount - for ODBC_STORAGE needs to be fixed to support multiple mailboxes - - * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for - ODBC_STORAGE. Also, leave a comment for this to be fixed for - IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was - working on this code right now for another reason. This is broken - even worse in trunk, but for a different reason. The fact that - the mailbox option supported multiple mailboxes is completely not - obvious from the code in the channel drivers. Anyway, I will fix - that in another commit ... - - * apps/app_meetme.c: Fix a problem with the combination of the 'F' - option to pass DTMF through a conference and options that use - DTMF to activate various features. The problem was that the BEGIN - frame would be passed through, but the END frame would get - intercepted to activate a feature. Then, the other conference - members would hear DTMF for forever, which they didn't seem to - like very much. (closes issue #10400, reported by stevefeinstein, - fixed by me) - -2007-08-08 19:29 +0000 [r78646] Jason Parker <jparker@digium.com> - - * doc/jabber.txt: Fix mogs email address. - -2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixed some compiler warnings so that - compiling with dev-mode and IMAP storage would not have any - errors. This section of code may get changed again shortly since - my change uncovers a rather silly bit of logic. - - * apps/app_queue.c: Changing a bit of logic so that someone will - NEVER exit the queue on timeout unless they have enabled the 'n' - option. This commit relates to issue #10320. Thanks to - jfitzgibbon for detailing the idea behind this code change. - -2007-08-08 13:51 +0000 [r78569] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample: (closes issue #10335) Reported by: - adamgundy Update sip.conf to include another scenario where - directrtpsetup will fail. - -2007-08-07 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.10 released. - -2007-08-07 20:57 +0000 [r78488] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c: Fix the build of this module on 64-bit - platforms - -2007-08-07 19:43 +0000 [r78450] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: The logic behind inboxcount's return value - was reversed in has_voicemail and message_count. (closes issue - #10401, reported by st1710, patched by me) - -2007-08-07 19:34 +0000 [r78437] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Don't free the environment handle when the - connection fails, because other connections might be depending - upon it - -2007-08-07 19:11 +0000 [r78416] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Allow chan_sip to build in devmode - -2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, res/res_config_odbc.c, - apps/app_directory.c: Reconnection doesn't happen automatically - when a DB goes down (fixes issue #9389) - -2007-08-07 18:25 +0000 [r78375] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Properly check the capabilities count to - avoid a segfault. (ASA-2007-019) - -2007-08-07 17:45 +0000 [r78371] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) | - 4 lines Revert patch committed for issue #9660. It broke E&M - trunks. (closes issue #10360) (closes issue #10364) ........ - -2007-08-06 21:41 +0000 [r78275] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Add additional DTMF log messages to help when - debugging issues. - -2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix an issue where dynamic threads can get - free'd, but still exist in the dynamic thread list. (closes issue - #10392, patch from Mihai, with credit to his colleague, Pete) - - * include/asterisk/linkedlists.h: Fix the return value of - AST_LIST_REMOVE(). This shouldn't be causing any problems, - though, because the only code that uses the return value only - checks to see if it is NULL. (closes issue #10390, pointed out by - mihai) - -2007-08-06 16:32 +0000 [r78182] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: It is possible for a transfer to occur - before the remote device has our tag in which case they send none - in the transfer. In this case we need to not fail the transfer - dialog lookup. - -2007-08-06 16:30 +0000 [r78180] Jason Parker <jparker@digium.com> - - * main/config.c: Fix an issue with using UpdateConfig (manager - action) where escaped semicolons in a config would be converted - to just semicolons (\; to ;) Issue 9938 - -2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that - we pass through RTP timestamp information we need to make the - allowed timestamp skew considerably less. There are situations - where a source may change and due to the timestamp difference the - receiver will experience an audio gap since we did not indicate - by setting the marker bit that the source changed. - - * configure, configure.ac: (closes issue #10383) Reported by: rizzo - Include stdlib.h so NULL gets defined for gethostbyname_r checks. - -2007-08-06 13:33 +0000 [r78164] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fixed a mistake I made in realtime_peer - which caused it to return NULL every time. Thanks to Jon Fealy - for emailing me the correction. - -2007-08-05 14:18 +0000 [r78146] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes - bug #10382) - -2007-08-05 04:15 +0000 [r78143] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: Fix compilation failure when - MALLOC_DEBUG is enabled, but DEBUG_THREADS is not - -2007-08-05 03:29 +0000 [r78139] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: If peer is not found, the error message is - misleading (should be peer not found, not ACL failure) - -2007-08-03 20:25 +0000 [r78103] Mark Michelson <mmichelson@digium.com> - - * main/config.c, channels/chan_sip.c, include/asterisk/config.h: - Changed the behavior of sip's realtime_peer function to match the - corresponding way of matching for non-realtime peers. Now matches - are made on both the IP address and port number, or if the - insecure setting is set to "port" then just match on the IP - address. In order to accomplish this, I also added a new API - call, ast_category_root, which returns the first variable of an - ast_category struct - -2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: (closes issue #10194) Reported by: - blitzrage Patches: bug0010194 uploaded by vovochka Tested by: - blitzrage Fix a problem when you call Voicemail() with multiple - mailboxes specified and ODBC_STORAGE is in use. The audio part of - the message was only given to the first mailbox specified. - - * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some - improvements to lock debugging. These changes take effect with - DEBUG_THREADS enabled and provide the following: * This will keep - track of which locks are held by which thread as well as which - lock a thread is waiting for in a thread-local data structure. A - reference to this structure is available on the stack in the - dummy_start() function, which is the common entry point for all - threads. This information can be easily retrieved using gdb if - you switch to the dummy_start() stack frame of any thread and - print the contents of the lock_info variable. * All of the - thread-local structures for keeping track of this lock - information are also stored in a list so that the information can - be dumped to the CLI using the "core show locks" CLI command. - This introduces a little bit of a performance hit as it requires - additional underlying locking operations inside of every - lock/unlock on an ast_mutex. However, the benefits of having this - information available at the CLI is huge, especially considering - this is only done in DEBUG_THREADS mode. It means that in most - cases where we debug deadlocks, we no longer have to request - access to the machine to analyze the contents of ast_mutex_t - structures. We can now just ask them to get the output of "core - show locks", which gives us all of the information we needed in - most cases. I also had to make some additional changes to astmm.c - to make this work when both MALLOC_DEBUG and DEBUG_THREADS are - enabled. I disabled tracking of one of the locks in astmm.c - because it gets used inside the replacement memory allocation - routines, and the lock tracking code allocates memory. This - caused infinite recursion. - - * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control - frames when the mohinterpret option is set to "passthrough". This - was pointed out by Kevin in the middle of a training session. - - * channels/chan_iax2.c: Don't reuse the timespec that was set to 0 - in the previous timedwait as it will just return immediately. - Also, fix some logic so the thread's lock isn't unlocked twice in - the weird case of dynamic threads getting acquired right after a - timeout. (pointed out by SteveK) - -2007-08-02 21:53 +0000 [r77993-77996] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we - actually allow 6 chars to be sent. Also make note of the "A" - option of date format. Issue 9779, modifications by DEA, wedhorn, - and myself. - - * channels/chan_skinny.c: If a device disconnects, the session will - go away. If this happens during call setup, we need to give up. - Issue 10325. - -2007-08-02 19:25 +0000 [r77949] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix the case where a dynamic thread times - out waiting for something to do during the first time it runs. - This shouldn't ever happen, but we should account for it anyway. - (pointed out by pete, who works with mihai) - -2007-08-02 18:42 +0000 [r77947] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure we clear the prompt status - message on a hangup. Also rearrange messages to better fit with - what a wireshark trace shows it should be. Issue 10299, initial - patch and solution by sbisker, modified by me to fit with - wireshark trace. - -2007-08-02 18:21 +0000 [r77945] Steve Murphy <murf@digium.com> - - * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 - line This patch hopefully solves 10141; The user is running with - it, and it doesn't appear to harm asterisk's operation, and may - prevent a crash. I'll store it in 1.2, as we have shut down - support on 1.2, but since I developed the patch before support - finished, and it might affect 1.4 and trunk, I'm going ahead with - it. ........ - -2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix another race condition in the handling - of dynamic threads. If the dynamic thread timed out waiting for - something to do, but was acquired to perform an action - immediately afterwords, then wait on the condition again to give - the other thread a chance to finish setting up the data for what - action this thread should perform. Otherwise, if it immediately - continues, it will perform the wrong action. (reported on IRC by - mihai, patch by me) (related to issue #10289) - - * channels/chan_iax2.c: Add another sanity check to - vnak_retransmit(). This check ensures that frames that have - already been marked for deletion don't get retransmitted. (closes - issue #10361, patch from mihai) - -2007-08-02 15:15 +0000 [r77890-77894] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure that we show the correct - extension if dialed from a macro "From: 5555" rather than "From: - s" Issue 10358, initial patch by DEA, reworked by me to use S_OR, - tested by sbisker - - * channels/chan_skinny.c: Put in some additional debug information - for softkey/stimulus messages. Issue 10291, patch by DEA. - -2007-08-01 22:16 +0000 [r77887] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix some race conditions which have been - causing weird problems in chan_iax2. The most notable problem is - that people have been seeing storms of VNAK frames being sent due - to really old frames mysteriously being in the retransmission - queue and never getting removed. It was possible that a dynamic - thread got created, but did not acquire its lock before the - thread that created it signals it to perform an action. When this - happens, the thread will sleep until it hits a timeout, and then - get destroyed. So, the action never gets performed and in some - cases, means a frame doesn't get transmitted and never gets freed - since the scheduler never gets a chance to reschedule - transmission. Another less severe race condition is in the - handling of a timeout for a dynamic thread. It was possible for - it to be acquired to perform at action at the same time that it - hit a timeout. When this occurs, whatever action it was acquired - for would never get performed. (patch contributed by Mihai and - SteveK) (closes issue #10289) (closes issue #10248) (closes issue - #10232) (possibly related to issue #10359) - -2007-08-01 22:14 +0000 [r77886] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does - not compile cleanly (missing def) - -2007-08-01 21:08 +0000 [r77883] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an issue that caused one-way audio on - some newer devices (specifically the 7921), due to sending - packets in the wrong order during hangup. Also make sure we clear - tones/messages on the correct line/instance. Issue 10291, patch - by DEA, tested by sbisker and myself. - -2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp <jcolp@digium.com> - - * main/cli.c: (closes issue #10351) Reported by: ftarz Some - platforms don't like it when you pass NULL to vsnprintf so pass - "" instead. - - * include/asterisk/threadstorage.h, channels/chan_mgcp.c, - apps/app_voicemail.c, main/acl.c, utils/smsq.c, - channels/chan_iax2.c: Add some fixes for building on Solaris. - - * main/utils.c: Whoops, I meant R_5 not R5. - - * configure, configure.ac: And for my last trick... make sure that - if gethostbyname_r is exported by a library that it is used. - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/utils.c: Extend autoconf logic to determine which version of - gethostbyname_r is on the system. - -2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixes an issue I introduced to queues wherein a - queue with joinempty=yes would kick people out of the queue - because of erroneously thinking the 'n' option was in use. - (closes issue #10320, reported by jfitzgibbon, patched by me, - tested by blitzrage and me) Thank you blitzrage for all the - testing you've done lately with queues! It's much appreciated! - - * apps/app_queue.c: If a queue uses dynamic realtime members, then - the member list should be updated after each attempt to call the - queue. This fixes an issue where if a caller calls into a queue - where no one is logged in, they would wait forever even if a - member logged in at some point. (closes issue #10346, reported by - and tested by blitzrage, patched by me) - -2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon <telesistant@hotmail.com> - - * apps/app_rpt.c: Much newer version, 0.70 with much additions - - * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF - receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in - logic in TONE_VERIFY/RELAX mode in chan_zap. - -2007-07-31 20:59 +0000 [r77844] Steve Murphy <murf@digium.com> - - * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1 - line This probably isn't super-general, but it's a first stab at - using kill -11 to generate a core file instead of gcore. ........ - -2007-07-31 16:17 +0000 [r77831] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c, include/asterisk/speech.h: Add a flag to the - speech API that allows an engine to set whether it received - results or not. - -2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without - DEBUG_THREADS or it does nothing - -2007-07-31 15:21 +0000 [r77824] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: This patch makes Asterisk send 100 Trying - provisional responses upon receipt of re-invites. This makes it - so that if there are two or more Asterisk servers between - endpoints, the Asterisk servers will not keep retransmitting the - re-invites. (closes issue #10274, reported by cstadlmann, patched - by me with approval from file) - -2007-07-30 20:17 +0000 [r77795] Jason Parker <jparker@digium.com> - - * main/say.c: Applications like SayAlpha() should not hang up the - channel if you request an "unknown" character such as a comma. - Instead, skip the character and move on. Issue 10083, initial - patch by jsmith, modified by me. - -2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix an issue that could potentially cause - corruption of the global iax frame queue. In the network_thread() - loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE - macro. However, to remove an element of the list within this - loop, it used AST_LIST_REMOVE, instead of - AST_LIST_REMOVE_CURRENT, which I believe could leave some of the - internal variables of the SAFE macro invalid. Mihai says that he - already made this change in his local copy and it didn't help his - VNAK storm issues, but I still think it's wrong. :) - - * res/res_agi.c: (closes issue #10279) Reported by: seanbright - Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by - seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch - uploaded by seanbright (license 71) Allow the "agi_network: yes" - line to be printed out in the AGI debug output. Also, allow - partial writes to be handled when writing out this line just like - it is for all of the others. - - * main/channel.c: file and I both committed changes for issue - #10301. Remove a duplicated assignment to restore the original - value of the previous channel. - -2007-07-30 18:43 +0000 [r77783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, res/res_agi.c: Merged revisions 77782 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) - | 2 lines Revert change in revision 71656, even though it fixed a - bug, because many people were depending upon the (broken) - behavior. ........ - -2007-07-30 17:29 +0000 [r77780] Russell Bryant <russell@digium.com> - - * main/channel.c: (closes issue #10301) Reported by: fnordian - Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian - (license 110) Additional changes by me Fix some problems in - channel_find_locked() which can cause an infinite loop. The - reference to the previous channel is set to NULL in some cases. - These changes ensure that the reference to the previous channel - gets restored before needing it again. I'm not convinced that the - code that is setting it to NULL is really the right thing to do. - However, I am making these changes to fix the obvious problem and - just leaving an XXX comment that it needs a better explanation - that what is there now. - -2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (closes issue #10327) Reported by: kkiely - Instead of directly mucking with the extension/context/priority - of the channel we are transferring when it has a PBX simply call - ast_async_goto on it. This will ensure that the channel gets - handled properly and sent to the right place. - - * main/channel.c: (closes issue #10301) Reported by: fnordian - Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian - (license 110) Restore previous behavior where if we failed to - lock the channel we wanted we would return to exactly the same - point as if we had just reentered the function. - - * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 - lines (closes issue #10334) Reported by: ramonpeek Pass through - the return value from macro_exec through the MacroIf application. - ........ - -2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Missing newline - -2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois - Patches: cdr_pgsql.patch uploaded by prashant (license 114) - Finish the Postgresql connection after the log messages are - printed so we don't access invalid memory. - - * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm - Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by - julianjm (license 99) Clear ONHOLD flag when decrementing the - onHold peer count. If we did not do this the count may keep - decreasing. - -2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: "re-invite" was misspelled - -2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp@digium.com> - - * main/channel.c: (closes issue #10302) Reported by: litnialex If a - DTMF end frame comes from a channel without a begin and it is - going to a technology that only accepts end frames (aka INFO) - then use the minimum DTMF duration if one is not in the frame - already. - -2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming@digium.com> - - * doc/mp3.txt: change protocol for downloads as well - - * doc/mp3.txt, sounds/Makefile: use new canonical name for download - server - -2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell@digium.com> - - * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined - in trunk, but not in 1.4. When Asterisk is compiled under dev - mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to - define it in the same way that trunk does. Also, revert the - change that added this define in the Makefile The advantage to - doing it this way is that buildopts.h gets installed when you - install Asterisk. Then, when building any out of tree modules, or - building asterisk-addons, these modules know which options the - rest of Asterisk was built with. - -2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson@digium.com> - - * Makefile, main/logger.c: Fixes to get ast_backtrace working - properly. The AST_DEVMODE macro was never defined so the majority - of ast_backtrace never attempted compilation. The makefile now - defines AST_DEVMODE if configure was run with --enable-dev-mode. - Also, changes were made to acccomodate 64 bit systems in - ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for - their roles in allowing me to get this committed - -2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/logger.c: Missed one - - * main/logger.c: Oops, that builtin define should be all-lowercase. - -2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson@digium.com> - - * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur, - meaning free gets called on the same variable twice. This patch - sets the connection to NULL after calls to PQfinish so that the - problem does not occur. Also in this patch, prashant_jois - informed me that it is safe to pass a null pointer to PQfinish, - so I have removed the check for conn's existence from - my_unload_module. (closes issue 10295, reported by junky, patched - by me with input from prashant_jois) - -2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf@digium.com> - - * apps/app_meetme.c: This fix solves problem with intense squelch - noise when someone joins conf in bug 9430; We repro'd the problem - with meetme opts of 'CciMo'; Josh Colp supplied this patch, and - I'm applying it. It looks like playing the recorded username will - louse up the next thing played into the channel. Josh rearranged - the code so as to start things over before playing data directly - into the conference. - -2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd - Add SPEECH_DTMF_TERMINATOR variable so the user can specify the - digit to terminate a DTMF string with. If none is specified then - no terminator will be used. - -2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: chan->emulate_dtmf_duration is an unsigned int, - not a signed int, so use %u instead of %d in the format string - -2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker@digium.com> - - * /: so are my fingers... - - * /: autotagexternals script is still obviously misbehaving... - - * /: use autotagged externals - -2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp@digium.com> - - * configure, acinclude.m4: Fix autoconf logic for finding OpenH323 - when it is not in the first place searched (/usr/share/openh323). - -2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo@icir.org> - - * main/rtp.c: set the sequence number in a frame for all frame - types - -2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 - line this fixes bug 10293, where the error message because - defaultzone or loadzone was not defined was confusing ........ - -2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 - Jul 2007) | 2 lines Oops, res contains the error code, not errno. - I was wondering why a mutex was reporting "No such file or - directory"... ........ - - * main/app.c: Found another place where we should be using the - umask (thanks jcmoore) - -2007-07-24 Jason Parker <jparker@digium.com> - - * Asterisk 1.4.9 released. - -2007-07-24 16:42 +0000 [r76803-76805] 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) - -2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Added a membercount variable to call_queue - struct which keeps track of the number of logged in members in a - particular queue. This makes it so that the 'n' option for - Queue() can act properly depending on which strategy is used. If - the strategy is roundrobin, rrmemory, or ringall, we want to ring - each phone once before moving on in the dialplan. However, if any - other strategy is used, we will only ring one phone since it - cannot be guaranteed that a different phone will ring on - subsequent attempts to ring a phone. As a side effect of this, - the QUEUE_MEMBER_COUNT dialplan function now just reads the - membercount variable instead of traversing through the member - list to figure out how many members there are. Special thanks to - blitzrage for helping to test this out. (closes issue #10127, - reported by bcnit, patched by me, tested by blitzrage) - -2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: It was our stated intention for 1.4 that - files created in app_voicemail should depend upon the umask. - Unfortunately, mkstemp() creates files with mode 0600, regardless - of the umask. This corrects that deficiency. - -2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix some incorrect softkey labels in - messages. Don't try to play dialtone in some unimplemented - features. - -2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul - 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not - force channel format changes when a generator is present. The - generator may have changed the formats itself and changing them - back would cause issues. ........ - -2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Don't try to queue up hold/unhold frames - on a non-existent channel. Issue 10276. - -2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp@digium.com> - - * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux - by putting the value of the digit char in an int. - - * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 - lines (closes issue #10236) Reported by: homesick Patches: - rpid_1.4_75840.patch uploaded by homesick (license 91) Accept - Remote Party ID on guest calls. ........ - - * channels/chan_skinny.c: (closes issue #10268) Reported by: - mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak - (license 7) Add another OS that has to use the Macros for byte - ordering. - -2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Use a signed integer for storing the number - of bytes in the packet read from the network. Using an unsigned - value here made it impossible to handle an error returned from - recvfrom(). Furthermore, in the case that recvfrom() did return - an error, this would cause a crash due to a heap overflow. - (closes issue #10265, reported by and fix suggested by - timrobbins) - -2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) | - 4 lines Backport a fix for a memory leak that was fixed in trunk - in reivision 76221 by rizzo. The memory used for the localaddr - list was not freed during a configuration reload. ........ - -2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf@digium.com> - - * sounds/Makefile: This patch from 10249 is worth applying! It - prevents downloading sound files if they are already downloaded. - Darn Practical, if you ask me - -2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Allow getting a call from an existing - "sub" channel. Cancel ringing if endpoint hangs up before - answering. Fixes were backported from trunk (there was apparently - a bit of confusion during merge of a previous patch). (closes - issue #10241) - - * main/manager.c: Eliminate a compiler warning with gcc 4.2 by - constifying a char * - - * channels/chan_skinny.c: It's possible for sub->owner to be NULL - here if you cancel the call immediately after/during sending a - digit. - -2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson@digium.com> - - * apps/app_directory.c: When using users.conf for the entries in - the directory, if multiple users had the same last name, only the - first user listed would be available in the directory. (closes - issue #10200, reported by mrskippy, patched by me) - -2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell@digium.com> - - * main/channel.c: Use the define that specifies the default length - of an artificially created DTMF digit in the ast_senddigit() - function. The define is set to 100ms by default, which is the - same thing that this function was using. But, using the define - lets changes take effect in this case, as well as the others - where it was already used. - -2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 - lines (closes issue #10247) Reported by: fkasumovic Patches: - chan_sip.patch uploaded by fkasumovic (license #101) Drop any - peer realm authentication entries when reloading so multiple - entries do not get added to the peer. ........ - - * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic - Patches: res_conver.patch uploaded by fkasumovic (license #101) - Use the last occurance of . to find the extension, not the first - occurance. - - * apps/app_queue.c: Move makeannouncement variable declaration to - proper place. - -2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Remove some duplicate code. - -2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: The diff on this looks pretty big but all I did - was remove a pointless if statement (always evaluates true). - - * apps/app_queue.c: Changes in handling return values of several - functions in app_queue. This all started as a fix for issue - #10008 but now includes all of the following changes: 1. - Simplifying the code to handle positive return values from ast - API calls. 2. Removing the background_file function. 3. The fix - for issue #10008 (closes issue #10008, reported and patched by - dimas) - -2007-07-19 15:53 +0000 [r75928] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) | - 6 lines When processing full frames, take sequence number - wraparound into account when deciding whether or not we need to - request retransmissions by sending a VNAK. This code could cause - VNAKs to be sent erroneously in some cases, and to not be sent in - other cases when it should have been. (closes issue #10237, - reported and patched by mihai) ........ - -2007-07-18 22:59 +0000 [r75807] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Need to make sure we set milliseconds and - timestamp - pointed out by the recent ast_ time stuff from - Tilghman - -2007-07-18 21:09 +0000 [r75759] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | - 5 lines When traversing the queue of frames for possible - retransmission after receiving a VNAK, handle sequence number - wraparound so that all frames that should be retransmitted - actually do get retransmitted. (issue #10227, reported and - patched by mihai) ........ - -2007-07-18 20:40 +0000 [r75749] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007) - | 2 lines Store prior to copy (closes issue #10193) ........ - -2007-07-18 20:17 +0000 [r75732] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Umm, why are we transmitting dialtone on - cfwdall? - -2007-07-18 20:00 +0000 [r75712] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c, - pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these - Asterisk won't build under devmode using GCC 4.2. - -2007-07-18 19:54 +0000 [r75707-75711] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fixes for 7935/7936 conference phones. - Issue 9245, patch by slimey. - - * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue - 9887, patches by DEA - -2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007) - | 1 line removed the word 'pissed' from ast_log(...) function - call for BE-90 ........ - -2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Few more places that needs to check for - onhold state. - - * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar - It is possible for hold status to exist without call limits set, - so we need to ensure update_call_counter is executed regardless. - - * channels/chan_h323.c: Don't bother reloading chan_h323 if it did - not load successfully in the first place. This would otherwise - cause a crash. - - * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record - the threadid of each running thread before shutting them down as - the thread themselves may change the value. - -2007-07-18 12:29 +0000 [r75529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_meetme.c: Using a freed frame causes crashes (closes - issue #9317) - -2007-07-17 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.8 released. - -2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant <russell@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 - Jul 2007) | 3 lines Properly check for the length in the skinny - packet to prevent an invalid memcpy. (ASA-2007-016) ........ - - * main/rtp.c: cast arguments to ast_log so that it builds without - warnings for me - - * channels/iax2-parser.c, channels/iax2-parser.h, /, - channels/chan_iax2.c: Merged revisions 75444 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | - 5 lines Ensure that when encoding the contents of an ast_frame - into an iax_frame, that the size of the destination buffer is - known in the iax_frame so that code won't write past the end of - the allocated buffer when sending outgoing frames. (ASA-2007-014) - ........ - - * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | - 4 lines After parsing information elements in IAX frames, set the - data length to zero, so that code later on does not think it has - data to copy. (ASA-2007-015) ........ - -2007-07-17 20:40 +0000 [r75439] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Ensure that the pointer to STUN data does not go to - unaccessible memory. (ASA-2007-017) - -2007-07-17 20:33 +0000 [r75437] Russell Bryant <russell@digium.com> - - * res/res_agi.c: (issue #10210) Reported by: juggie Patches: - 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by: - juggie, blitzrage Log a warning if someone uses DeadAGI on a live - channel. - -2007-07-17 20:03 +0000 [r75405] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists - can return -1 on failure, so I need to be sure that we only enter - the if statement if it is successful. Related to my fix to issue - #10186 - -2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant <russell@digium.com> - - * main/pbx.c: (closes issue #10209) Reported by: juggie Patches: - 10209-trunk-2.patch uploaded by juggie Tested by: juggie, - blitzrage In ast_pbx_run(), mark a channel as hung up after an - application returned -1, or when it runs out of extensions to - execute. This is so that code can detect that this channel has - been hung up for things like making sure DeadAGI is used on - actual dead channels, and is beneficial for other things, like - making sure someone doesn't try to start spying on a channel that - is about to go away. - - * res/res_agi.c: Remove a duplicated newline character in AGI debug - output. (closes issue #10207, patch by seanbright) - -2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming <kpfleming@digium.com> - - * main/dns.c, /: Merged revisions 75304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007) - | 3 lines provide proper copyright/license attribution for this - structure that was copied from a BSD-licensed header file long, - long ago... ........ - - * /: another fix that is not needed here (finishing up 75251) - -2007-07-16 18:16 +0000 [r75253] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: Restoring functionality from 1.2 wherein - Retrydial will not exit if there is no announce file specified. - This change makes it so that if there is no announce file - specified, the application will continue until finished (or - caller hangs up). If a bogus announce file is specified, then a - warning message will be printed saying that the file could not be - found, but execution will still continue. (closes issue #10186, - reported by jon, patched by me) - -2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming <kpfleming@digium.com> - - * /: block change that is not relevant here - -2007-07-13 20:36 +0000 [r75108] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13 - Jul 2007) | 3 lines Fix a couple potential minor memory leaks. - load_moh_classes() could return without destroying the loaded - configuration. ........ - -2007-07-13 20:15 +0000 [r75078] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul - 2007) | 5 lines Fixed an issue where chanspy flags were - uninitialized if no options were passed. What triggered this - investigation was an IRC chat where some people's quiet flags - were set while others' weren't even though none of them had - specified the q option. ........ - -2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 - Jul 2007) | 6 lines Ensure that adding a user to the list of - users of a specific music on hold class is not done at the same - time as any of the other operations on this list to prevent list - corruption. Using the global moh_data lock for this is not ideal, - but it is what is used to protect these lists everywhere else in - the module, and I am only changing what is necessary to fix the - bug. ........ - - * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | - 12 lines (closes issue #9660) Reported by: mmacvicar Patches - submitted by: bbryant, russell Tested by: mmacvicar, marco, - arcivanov, jmhunter, explidous When using a TDM400P (and probably - other analog cards) there was a chance that you could hang up and - pick the phone back up where it has been long enough to be not - considered a flash hook, but too soon such that the device - reports that it is busy and the person on the phone will only - hear silence. This patch makes chan_zap more tolerant of this and - gives the device a couple of seconds to succeed so the person on - the phone happily gets their dialtone. ........ - -2007-07-12 23:00 +0000 [r74998] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Change to my previous fix regarding agent - logoff soft. Now uses deferlogoff instead of loginstart since - loginstart is used after logoff. Thanks to makoto for pointing - this out and suggesting the fix. (closes issue #10178, reported - and patched by makoto, with modification by me) - -2007-07-12 20:42 +0000 [r74955] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: This patch resolves 10143; thanks to irroot - for the patch; looked acceptable. Let the community decide if it - messes things up - -2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Whoops... didn't want this to be returned to 0 - each iteration. - - * main/channel.c: When waiting for a digit ensure that a begin - frame was received with it, not just an end frame. (issue #10084 - reported by rushowr) - -2007-07-12 16:53 +0000 [r74839-74866] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: It helps if I actually add this stuff for - the 7921 too - otherwise it won't actually do much of anything. - - * channels/chan_skinny.c: Add device ID for 7921 wireless skinny - phone - - * channels/chan_skinny.c: Fix dialing in skinny that was broken in - some cases. Issue 10136, fix provided by DEA. - -2007-07-12 15:53 +0000 [r74815] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul - 2007) | 2 lines Only print out a warning for situations where it - is actually helpful. (issue #10187 reported by denke) ........ - -2007-07-11 22:57 +0000 [r74767] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | - 5 lines The function make_trunk() can fail and return -1 instead - of a valid new call number. Fix the uses of this function to - handle this instead of treating it as the new call number. This - would cause a deadlock and memory corruption. (possible cause of - issue #9614 and others, patch by me) ........ - -2007-07-11 21:14 +0000 [r74722] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 - Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft" - did not work...at all. Now it does. (closes issue #10178, - reported and patched by makoto, with slight modification for 1.4 - and trunk by me) ........ - -2007-07-11 18:34 +0000 [r74657] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11 - Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows - the condition that uses LIKE. This fixes realtime extensions with - ODBC. (closes issue #10175, reported by stuarth, patch by me) - ........ - -2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy <murf@digium.com> - - * Makefile: This fixes 10172, where the entire man8 dir gets - removed during an uninstall of asterisk - - * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt: - further reversion of previously applied floating point stuff for - expr2 - -2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Instead of - figuring out kernel versions that have compiler.h and not... - let's just use autoconf to check for it's presence. (issue #10174 - reported by francesco_r) - - * channels/chan_phone.c: Only check if we need to do a SIGMA based - tone generation if we have a card. (issue #10179 reported by - mikowhy) - -2007-07-10 23:32 +0000 [r74476] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Forwarding a message with IMAP storage was - storing the message in the sender's box instead of the forwarded - mailbox. (closes issue #10138, reported and patched by jaroth) - -2007-07-10 19:58 +0000 [r74374-74428] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 - lines Fix an issue where it was possible to have a service level - of over 100% Between the time recalc_holdtime and update_queue - was called, it was possible that the call could have been hungup. - Move both additions to the same place, so this won't happen. - Issue 10158, initial patch by makoto, modified by me. ........ - - * main/dns.c: Don't use #if to check if something is defined - use - #ifdef instead. Pointed out by kpfleming - - * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul - 2007) | 4 lines Fix an issue with wrapuptime not working when - using AgentLogin. Issue 10169, patch by makoto, with a minor mod - by me to not re-break issue 9618 ........ - - * main/dns.c, /, configure, include/asterisk/autoconfig.h.in, - configure.ac: Merged revisions 74373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 - lines Use res_ndestroy on systems that have it. Otherwise, use - res_nclose. This prevents a memleak on NetBSD - and possibly - others. Issue 10133, patch by me, reported and tested by scw - ........ - -2007-07-10 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.7.1 released. - -2007-07-10 16:00 +0000 [r74323] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: fix an uninitialized variable - -2007-07-10 15:38 +0000 [r74317] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 - lines Fix a small typo in description in of Voicemail() - application. Issue 10170, patch by casper. ........ - -2007-07-10 15:31 +0000 [r74314] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 - Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue - #10075, this part reported by jmls on IRC, patch by me) ........ - -2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp <jcolp@digium.com> - - * /, main/app.c: Merged revisions 74264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 - lines Ensure the group information category exists before trying - to do a string comparison with it. (issue #10171 reported by - mlegas) ........ - - * channels/chan_sip.c: Only spit out an inringing warning message - when it is applicable. Since call limits are already toast in - realtime let's not scare the user if they are using it. (issue - #10166 reported by bcnit) - -2007-07-09 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.7 released. - -2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Update the configure script to check for - a required function that is not present in the 1.2 version of - libpri. This will prevent the configure script from thinking that - it has compatible libpri support for Asterisk 1.4, when it - actually does not because the installed version is from 1.2. - - * res/res_musiconhold.c: (closes issue #10123) Reported by: - blitzrage Patches submitted by: juggie, qwell, me Tested by: - blitzrage When trying to find a music on hold class to use, try - all of the options, instead of only the first one that is set. - Also, change the MusicOnHold applications to not hang up on the - channel when a class can not be found. - -2007-07-09 20:19 +0000 [r74159] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 - lines Several chan_zap options were not working on reload because - they were arbitrarily disallowed when reloading some/most PRI - options (such as signalling) was disallowed. Options such as - polarityonanswerdelay and answeronpolarityswitch can safely be - changed on a reload. This corrects that behavior. Issue 9186, - patch by tzafrir. ........ - -2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Forgot to get rid of an extraneous debug - message. - - * apps/app_queue.c: The n option for Queue should make the queue - exit immediately after failure to reach any members and should - not be dependent on the timeout value passed to Queue (closes - issue #10127, reported by bcnit, repaired by me) - -2007-07-09 15:32 +0000 [r74082] Joshua Colp <jcolp@digium.com> - - * channels/chan_skinny.c: Only destroy the scheduler context if it - was allocated. (issue #10124 reported by gzero) - -2007-07-09 14:57 +0000 [r74047] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixed a logic error in leave_voicemail. - Pass the mailbox instead of the context to inbox_count when the - context is "default." (closes issue #10135, reported by yannj, - repaired by me) - -2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp <jcolp@digium.com> - - * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread - synchronization tweaks. (issue #10124 reported by gzero) - - * configure, acinclude.m4: Use AC_CHECK_HEADER to check for - ptlib/openh323 to allow for cross compiling. (issue #9675 - reported by zandbelt) - -2007-07-09 04:03 +0000 [r73985] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while - 'make progdocs'. (Closes issue #10104) - -2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp <jcolp@digium.com> - - * main/cdr.c: Give Agent channel names priority when doing CDR - merging. (issue #10011 reported by krtorio) - - * pbx/pbx_config.c: Add a few sanity checks when writing out the - dialplan. (issue #10157 reported by dome) - -2007-07-08 09:47 +0000 [r73849] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: While tracking down a bug, I need some more - history. Dumphistory is very useful, indeed. - -2007-07-06 23:02 +0000 [r73769] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | - 4 lines If a sip_pvt struct has already registered an extension - state callback, remove the old one before adding a new one. If - this isn't done, Asterisk will crash. (issue #10120) ........ - -2007-07-06 16:36 +0000 [r73727] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixing a rare case which causes voicemail - to crash when compiled with IMAP storage. inboxcount has the - possibility of finding an "interactive" vm_state when no - persistent "non-interactive" vm_state exists for that mailbox. If - this should happen when someone attempts to leave a message, it - results in a crash. This patch, along with my commit in revision - 72670 fix issue 10053, reported by jaroth. closes issue #10053 - -2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 - Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras - Patches submitted by: Corydon76 Tested by: apsaras Fix a problem - with MSSQL 2005 by explicitly stating that '\' is being used as - an escape character. ........ - - * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | - 7 lines (closes issue #10125) Reported by: makoto Patches - submitted by: makoto This fixes a crash in chan_sip that happens - when the bindaddr setting is not valid on Asterisk startup, gets - fixed, and then a reload gets issued. ........ - -2007-07-06 15:27 +0000 [r73675] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 - Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. - (issue 9618, reported by jiddings, patched by moi) closes issue - #9618 ........ - -2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant <russell@digium.com> - - * BUGS: fix a little spelling error - - * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop - the monitor thread if it was never started. (closes issue #10124, - reported by gzero, fixed by me) - - * channels/chan_iax2.c: copy from the correct buffer when deferring - a full frame (related to issue #9937) - - * channels/chan_iax2.c: * Store the call number that a thread is - processing without the full frame bit set to ease debugging * - When deferring a full frame for processing, stick it into the - queue for the thread that is processing frames for that call, not - the one that read the current frame and is about to go back into - the idle list (related to issue #9937) - -2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) - | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just - like we don't support it for G.729 ........ - -2007-07-05 20:50 +0000 [r73512] Russell Bryant <russell@digium.com> - - * res/res_features.c: Pass HOLD and UNHOLD frames to the other - channel when they are returned from a native bridge function. - This fixes a problem where when two zap channels are natively - bridged and one does a flash hook, the other channel did not - receive music on hold. (Reported to me directly by Doug Bailey at - Digium) - -2007-07-05 19:18 +0000 [r73467] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 - lines Copy language information to the dialog structure when - calling a peer for situations where a PBX may be started on the - dialed channel. (issue #10121 reported by clegall_proformatique) - ........ - -2007-07-05 15:59 +0000 [r73400] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Correcting a minor CLI bug I found. When - issuing the queue show command, if you type queue show and then - press tab, you can continue pressing tab and it will keep - auto-completing queue names even though only 1 queue can be used - as an argument. - -2007-07-05 15:28 +0000 [r73398] Russell Bryant <russell@digium.com> - - * channels/chan_vpb.cc, channels/Makefile: Make this module build - for me in dev-mode - -2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 - lines Tweak spy locking. (issue #9951 reported by welles) - ........ - - * channels/chan_local.c, /: Merged revisions 73318 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul - 2007) | 2 lines Actually check to make sure a PBX was started on - one of the Local channels instead of blindly assuming it was. - (issue #10112 reported by makoto) ........ - - * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 - lines Reset ServicelevelPerf variable back to 0 if we are unable - to calculate it each time... otherwise we will get previous - values. (issue #10117 reported by noriyuki) ........ - -2007-07-04 14:53 +0000 [r73208-73253] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 - Jul 2007) | 1 line bchannel configurations like echocancel and - volume control, need to be setuped on inbound calls too. ........ - - * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 - Jul 2007) | 1 line bad bug in overlapdial case, we called - start_pbx multiple times, because the state wasn't changed.. - ........ - -2007-07-03 20:17 +0000 [r73143] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile, - main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing - expr floating patch from 1.4; too much of a behavior change. If - you want this fix, try trunk instead. bug 9508. - -2007-07-03 15:42 +0000 [r73104-73106] Jason Parker <jparker@digium.com> - - * /: What the heck. This should not have happened. - - * /: use autotagged externals - -2007-07-03 12:38 +0000 [r73053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) - | 2 lines RetryDial should accept a 0 argument, but it does not, - because atoi does not distinguish between 0 and error (closes - issue #10106) ........ - -2007-07-03 08:17 +0000 [r73005] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 - Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only - be called from mISDN Source channels.. #9449 ........ - -2007-07-02 20:16 +0000 [r72933] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput, - main/Makefile, main/ast_expr2.h, main/ast_expr2.y, - main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support - for floating point numbers added to ast_expr2 $\[...\] exprs. - Fixes bug 9508, where the expr code fails with fp numbers. The - MATH function returns fp numbers by default, so this fix is - considered necessary. - -2007-07-02 18:18 +0000 [r72926] Russell Bryant <russell@digium.com> - - * main/manager.c: Remove a bogus comment and add proper locking to - the handler function for the CLI command to show information on - manager actions. - -2007-07-02 14:32 +0000 [r72888] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Added additional DTMF debug messages for when - emulation occurs. - -2007-07-02 08:41 +0000 [r72850-72852] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 72585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | - 1 line check if the bchannel stack id is already used, if so - don't use it a second time. Also added a release_chan lock, so - that the same chan_list object cannot be freed twice. chan_misdn - does not crash anymore on heavy load with these changes. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 72099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | - 1 line simplified generation for dummy bchannels, also we mark - them as dummies, so they are not used later as real-bchannels, - optimized the RESTART mechanisms, we block a channel now on - cause:44, and send out a RESTART automatically, then on reception - of RESTART_ACKNOWLEDGE we unblock the channel again. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged - revisions 72087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | - 1 line simplified channel finding and locking a lot. removed - unnecessary #ifdefed areas. ........ - -2007-07-01 23:52 +0000 [r72806] Russell Bryant <russell@digium.com> - - * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) | - 5 lines When appending lines to call files to keep track of - retries, write a leading newline just in case the original call - file did not have a newline at the end. This fix is in response - to a problem I saw reported on the asterisk-users mailing list. - ........ - -2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Tweak the configure script so that error - output isn't spewed to the console when searching for GTK2 libs, - and they aren't found. - - * formats/format_pcm.c: give format_pcm a more concise destription - -2007-06-29 19:07 +0000 [r72665] Luigi Rizzo <rizzo@icir.org> - - * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for - absence of the function. This was already done in trunk. - -2007-06-29 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.6 released. - -2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp <jcolp@digium.com> - - * main/cdr.c: Minor change for older GCC versions. - - * Makefile, configure, configure.ac, makeopts.in: Backport fix for - GCC versions without support for declaration-after-statement. - -2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/manager.c: Issue 10055 - Change memory allocation to use the - heap for a command, since the output has the potential to - overflow the stack (as it did here) - - * res/res_jabber.c: Fix 1.4 breakage - -2007-06-28 19:44 +0000 [r72493] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in: regenerate the - configure script for rizzo - -2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo <rizzo@icir.org> - - * configure.ac: add a check for gethostbyname_r so we can simplify - the handling e.g. in utils.c Also add comments on a couple of - features which are not working on FreeBSD. All the above has been - already done in trunk so the merge must be blocked. Can someone - please regenerate ./configure ? - - * Makefile, channels/chan_zap.c, main/say.c: Add - -Wdeclaration-after-statement to AST_DEVMODE flags to catch - variable declarations in the middle of a block. Fix the few - instances of the above spotted out by the compiler. All of this - has been already done or is not applicable in trunk, so the merge - of this change will be blocked. - - * apps/app_meetme.c: cast a time_t so that it does not conflict - with the print format. This change was already done on trunk so - this change needs to be blocked from merging. - -2007-06-27 23:29 +0000 [r72383] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 72373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | - 3 lines Reinstating patch. This actually fixes the problem, - however I was running a development branch without it and - mistakenly thought it wasn't fixed. Fixes issue #10010, and - #9654: 100% CPU usage caused by an asterisk console losing it's - controlling terminal. ........ - -2007-06-27 23:25 +0000 [r72381] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun - 2007) | 2 lines Update documentation to clarify variable usage - with MixMonitor. (issue #9494 reported by netoguy) ........ - -2007-06-27 23:03 +0000 [r72335] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 72333 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) | - 2 lines Reverted changes for earlier revisions 72259 to 72261. - Issue #9654, #10010 ........ - -2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp <jcolp@digium.com> - - * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to - our list. Since these are dynamic payloads the other side - shouldn't care. (issue #9426 reported by irroot) - - * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 - lines Fix issue where queue log events might be missing. (issue - #7765 reported by mtryfoss) ........ - -2007-06-27 21:08 +0000 [r72272] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) | - 5 lines Fix a minor issue with parsing the priority number. You - could have as much whitespace as you want around a numeric - priority, but you couldn't have any whitespace around a special - priority like "n" or "hint". (issue #10039, reported by mitheloc, - fixed by me) ........ - -2007-06-27 20:46 +0000 [r72260] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 72259 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) | - 4 lines Fixes 100% load when controlling terminal disappears. - Issue #9654, #10010 ........ - -2007-06-27 20:25 +0000 [r72257] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 72256 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 - lines I may possibly get shot for doing this... but... defer CDR - processing until after the channel has been dealt with. This - should eliminate all of the issues with channels going funky - (SIP/PRI) when you are posting CDRs to a database that is either - slow or unavailable and do not want to enable batching. ........ - -2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: use the proper type for storing group number - bits so that if someone specifies 'group=42' it will actually - work instead of being silently ignored - -2007-06-27 18:40 +0000 [r72182-72185] Jason Parker <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 17:31 +0000 [r72148] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Make the ast_read_noaudio API call behave better - under circumstances where DTMF emulation was happening and a - generator was setup. (issue #10065 reported by stevefeinstein) - -2007-06-27 17:10 +0000 [r72125] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c: Don't modify a variable that we don't want - modified. Make a copy of it instead. Issue 10029, patch by - phsultan with slight modifications by me (to remove needless - casts). - -2007-06-27 16:34 +0000 [r72112] Russell Bryant <russell@digium.com> - - * main/rtp.c: Only output debug information related to RTCP - timestamps when RTCP debug is turned on (issue #10066, patch by - me) - -2007-06-27 07:58 +0000 [r72042] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | - 1 line for inbound TE calls, we setup the bchannel when we get - the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. - removed some #if 0 areas which weren't used anymore. ........ - r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | - 1 line isdn_lib.c didn't compile ........ - -2007-06-27 00:58 +0000 [r72006] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work. - -2007-06-26 23:02 +0000 [r71953] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removing a pointless line. This variable - was already set earlier and between then and this line, there is - no way that the values on the right side of the assignment could - have changed. - -2007-06-26 20:36 +0000 [r71915] Jason Parker <jparker@digium.com> - - * main/rtp.c: Don't dereference a pointer that may be NULL here. - Issue 10017. - -2007-06-26 19:00 +0000 [r71877] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: A few changes, the ultimate goal of which - is to keep better track of the number of messages that a mailbox - currently has. A description of the changes: 1. Changed the - "updated" field of the vm_state struct to act more as a binary - semaphore than a counting semaphore, since its current - implementation made the inboxcount function not work properly. - This change falls in line with a change made by UPenn with their - IMAP setup and helps to sync our changes with theirs. 2. - Eliminated some redundant calls to get_vm_state_by_mailbox inside - leave_voicemail 3. Use the play_folder variable to keep track of - the number of old and new messages in a mailbox as the messages - are deleted 4. Added an increment to the number of new messages - that was not there previously in the leave_voicemail function - -2007-06-26 15:47 +0000 [r71796] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixing bug where the authuser was - mistakenly pulled from the mailbox string instead of the IMAP - user. (closes issue 10054, reported and patched by jaroth) - -2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007) - | 2 lines Issue 10062 - Trying to move a message without - selecting one first results in memory corruption ........ - - * /, res/res_agi.c: Merged revisions 71656 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007) - | 2 lines Issue 10035 - handle_exec returns a result inconsistent - with all of the other AGI commands ........ - -2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Build a peer as well when hash323 is - enabled in users.conf (issue #9599 reported by asagage) - - * channels/chan_agent.c: Minor tweak for queueing up the unhold - frame... this will teach me to do bugs while half asleep. (issue - #10046 reported by dimas) - -2007-06-25 12:40 +0000 [r71519] Russell Bryant <russell@digium.com> - - * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue - #10048, Matti) - -2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 - lines Ignore other URIs after the first in a 300 Multiple Choice - response. (issue #10041 reported by homesick) ........ - - * main/cdr.c: Fix it so 1.4 actually compiles on my box. - - * channels/chan_agent.c: Check to make sure the channel pointer is - present before queueing up an unhold frame on it. (issue #10046 - reported by dimas) - -2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant <russell@digium.com> - - * build_tools/prep_tarball: Include the menuselect-tree file in - tarballs to make builds from tarballs a little bit faster - - * main/asterisk.c, /: Merged revisions 71358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) | - 2 lines Revert the patch from issue 9654 due to an unexpected - side effect ........ - -2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so - peer->cdr is what we really wanted to use - - * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 - Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to - be able to set variables to the empty string. ........ - -2007-06-23 03:29 +0000 [r71230] Steve Murphy <murf@digium.com> - - * main/cdr.c, res/res_features.c: This patch is meant to fix 8433; - where clid and src are lost via bridging. - -2007-06-22 22:44 +0000 [r71214] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 - Jun 2007) | 1 line fixed a bug that was introduced by copy and - paste in the last commit ..bchannels weren't cleaned properly. - ........ - -2007-06-22 15:38 +0000 [r71096-71123] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 70672 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | - 1 line we activate the bchannels in TE mode on incoming calls - only when we want to connect the call. ........ - - * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 - Jun 2007) | 1 line forgot one place .. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 - Jun 2007) | 1 line on receiption of cause:44 we mark the channel - as in use and inform the user about the situation, we need to - test the RESTART stuff then. Also shuffled the - empty_chan_in_stack function after the bchannel cleaning - functions, to avoid race conditions. ........ - - * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 - Jun 2007) | 1 line when we send out a SETUP, but get no response, - we should cleanup everything after reception of a hangup. - ........ - - * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | - 1 line restart indicator 0x80 is correct, at least that's what - libpri does. ........ - - * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 - Jun 2007) | 1 line if the bridged partner is mISDN too we should - not send dtmf tones, they are transmitted inband always ........ - - * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 - Jun 2007) | 1 line if we have already some digits, we just stop - the tones. ........ - -2007-06-22 15:00 +0000 [r71068] Jason Parker <jparker@digium.com> - - * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged - revisions 71065 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 - lines Fix a few silly usages of ast_playstream() - it only ever - returns 0... Issue 10035 ........ - -2007-06-22 14:53 +0000 [r71066] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 71064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | - 10 lines Fixed infinite loop when controlling terminal was lost - and return value of input function wasn't checked for errors. - This would cause 100% cpu to be taken up. (closes issue #9654, - issue #10010) Reported by: mnicholson, and eserra Idea for the - patch from mnicholson, patched by me ........ - -2007-06-22 14:10 +0000 [r71063] Steve Murphy <murf@digium.com> - - * main/cdr.c: My conditions for merging amaflags info was naive; - DOCUMENTATION is the default, although null is possible; theft of - user-settable fields is not good. Just copy them, leave them - alone. - -2007-06-22 03:14 +0000 [r71003] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a small typo which ... well ... - completely broke chan_iax2. oops! (issue #9937, patch by me) - -2007-06-21 22:34 +0000 [r70949] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 70948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 - line This little fix is in response to bug 10016, but may not - cure it. The code is wrong, clearly. In a situation where you set - the CDR's amaflags, and then ForkCDR, and then set the new CDR's - amaflags to some other value, you will see that all CDRs have had - their amaflags changed. This is not good. So I fixed it. ........ - -2007-06-21 21:40 +0000 [r70899] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 - lines Don't explode if the gain option is specified without a - value. (issue #9274 reported by mfarver) ........ - -2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Put the thread reading from the socket back - in the idle list if it deferred the processing of a full frame to - another thread - - * channels/chan_iax2.c: If a full frame is received while one of - the iax2 threads is in the middle of handling a full frame for - the same call, queue it up for processing by that same thread - later instead of dropping it. (issue #9937, patch by me) - -2007-06-21 20:19 +0000 [r70841] Steve Murphy <murf@digium.com> - - * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 - line it was pointed out that the cdr_custom config load could get - a lock, and under certain circumstances, would never release it. - I also noted that the situation where more than one mapping spec - was warned about, but did not ignore further mappings as it had - promised. I think I have fixed both situations. ........ - -2007-06-21 19:49 +0000 [r70808] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: When volgain is used don't leave a - temporary file behind. (Closes Issue 8514, Reported and patched - by ulogic, code reviewed by Jason Parker) - -2007-06-21 15:22 +0000 [r70727] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Do not Packet2Packet bridge if packetization settings - do not allow it. (issue #9117 reported by phsultan) - -2007-06-21 15:21 +0000 [r70726] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Remove a couple of duplicate unlocks - -2007-06-21 13:58 +0000 [r70677] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Fix building with ODBC storage enabled. - (issue #10025 reported by denisgalvao) - -2007-06-21 13:00 +0000 [r70656] Steve Murphy <murf@digium.com> - - * main/cdr.c: Via complaints aired in asterisk-users, I submit - these changes, which allow cdr updates to see macro - context/exten, whether hung up or not - -2007-06-20 23:32 +0000 [r70554-70612] Jason Parker <jparker@digium.com> - - * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql. - Issue 10020, patch by my, with credit to prashant_jois for - pointing out the problem. - - * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race - condition fix - - * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur - when reloading the module. Issue 10022, patch by me, with credit - to prashant_jois for finding the bug. - -2007-06-20 22:22 +0000 [r70552] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 - lines Don't overwrite the configured username setting upon a - REGISTER. (issue #8565 reported by jsmith) ........ - -2007-06-20 20:53 +0000 [r70494] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure we clear the previously dialed - number if it did not exist. Issue 9958. - -2007-06-20 19:29 +0000 [r70445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) - | 2 lines Issue 9997 - Timelimit times out the wrong channel - ........ - -2007-06-20 18:46 +0000 [r70397] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | - 5 lines Fix a problem where an established call would not be - properly disconnected when a PRI disconnect is received depending - on which cause code was received. (issue #9588, original patch by - softins, updated patch from jtexter3, and some additional - feedback from mhardeman) ........ - -2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, main/frame.c: Put the speex packetization values back - in but disable it when setting up the smoother. - - * main/frame.c: Don't do packetization/smoother stuff with speex, - it doesn't work. - -2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant <russell@digium.com> - - * contrib/scripts/ast_grab_core: don't delete the backtrace in - ast_grab_core - - * channels/chan_gtalk.c: Only attempt to queue a hangup on the - owner channel if it actually exists. (issue #9795, patch from - zandbelt) - -2007-06-19 18:23 +0000 [r70062] Steve Murphy <murf@digium.com> - - * main/channel.c, /: Merged revisions 70053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 - line This fixes 9246, where channel variables are not available - in the 'h' exten, on a 'ZOMBIE' channel. The fix is to - consolidate the channel variables during a masquerade, and then - copy the merged variables back onto the clone, so the zombie has - the same vars that the 'original' has. ........ - -2007-06-19 17:07 +0000 [r70003] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 69992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 - lines Handle the CC field in the RTP header. (issue #9384 - reported by DoodleHu) ........ - -2007-06-19 16:24 +0000 [r69987] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 69986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 - lines Update BRIDGEPEER variable if set to the new channel name - when a masquerade happens. (issue #9699 reported by dimas) - ........ - -2007-06-19 15:22 +0000 [r69944] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix a crash that could occur when handing - device state changes. When the state of a device changes, the - device state thread tells the extension state handling code that - it changed. Then, the extension state code calls the callback in - chan_sip so that it can update subscriptions to that extension. A - pointer to a sip_pvt structure is passed to this function as the - call which needs a NOTIFY sent. However, there was no locking - done to ensure that the pvt struct didn't disappear during this - process. (issue #9946, reported by tdonahue, patch by me, patch - updated to trunk to use the sip_pvt lock wrappers by eliel) - -2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 - lines Perform an extra hangup check just in case. (issue #9589 - reported by bcnit) ........ - - * /, res/res_features.c: Merged revisions 69846 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 - lines Add parked call extension AFTER the parking slot has been - announced, otherwise two threads will try to handle the same - channel and it will go kaboom. (issue #9191 reported by japple) - ........ - - * main/callerid.c: Fix for building on PowerPC under Linux. - -2007-06-18 19:48 +0000 [r69796] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Issue 10005 - Segfault with missing - arguments, plus fix a missing define for SIP INFO channels - -2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't count RTP timeout when involved in a - T38 fax session. (issue #9222 reported by ivoc) - - * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 - lines Set the peer name on the dialog to the one configured in - sip.conf and NOT the username to be used for authentication - attempts. (issue #9967 reported by achauvin) ........ - -2007-06-18 17:46 +0000 [r69744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007) - | 2 lines Issue 9998 - Remove SIG prefix, since it's not - supported by ksh ........ - -2007-06-18 16:51 +0000 [r69708] Joshua Colp <jcolp@digium.com> - - * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called - for the first time so that it does not needlessly spit out - changed messages when the host really didn't change. - -2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant <russell@digium.com> - - * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c, - build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, - include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c: - To prevent 92138749238754 more reports of "I have unixodbc - installed, but still can't build *_odbc.so!", check for ltdl - directly, instead of just listing it as another library to - include in the unixodbc check in the configure script. This also - makes ltdl show up as a dependency in menuselect so people know - what to go install. (related to issue #9989, patch by me) - - * build_tools/prep_moduledeps: Change the use of "echo -e" to - "printf". On systems where /bin/sh is not bash, most of the lines - in menuselect-tree were getting a "-e" at the beginning of every - line. I'm surprised nobody noticed this, but I think the XML - parser was being very nice and ignoring them. - -2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't defer the BYE till later on a transfer - when the transfer itself goes kaboom and has no hope of working. - - * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock - something we never locked, and better handle a specific scenario - with doing an attended transfer between two non-bridged calls. - -2007-06-18 15:46 +0000 [r69660] Russell Bryant <russell@digium.com> - - * Makefile: Tweak paths for BSD systems (issue #10001, stuarth) - -2007-06-18 13:55 +0000 [r69625] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix issue where it would be possible for the - negotiated codecs to get set back to nothing. (issue #9992 - reported by yehavi) - -2007-06-15 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.5 released. - -2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell@digium.com> - - * res/res_features.c: Fix a silly deadlock in res_features that I - found while debugging on one of blitzrage's test machines. It was - one of the situations where he was seeing hung channels, and may - be the cause of some of the reports from other people. (related - to issue #9235) - -2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Add support for setting the maximum - length of acceptable DTMF in SpeechBackground. - -2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated - "SUCCESS" for both an answer of the incoming call on the trunk, - or if the trunk reached its ring timeout. This patch changes the - variable to say "RINGTIMEOUT" in that case. (issue #9973, - reported by n00dle, patch by me) - -2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker@digium.com> - - * main/config.c, /: Merged revisions 69469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 - lines Fix an issue where the line number in an unterminated - comment block error message would show the wrong line number. - "Reported" to me on #asterisk (somebody posted an error message, - and I happened to catch it) ........ - - * sounds/Makefile: Update to latest versions of sound files. - -2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming@digium.com> - - * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, - cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c, - main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c, - apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c, - main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, - channels/chan_iax2.c: use ast_localtime() in every place - localtime_r() was being used - -2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell@digium.com> - - * main/say.c: Fix some problems with saying dates and times for the - "tw" langauge (issue #9964, ljmid) - -2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker@digium.com> - - * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun - 2007) | 4 lines Change a quite broken while loop to a for loop, - so "continue;" works as expected instead of eating 99% CPU... - Issue 9966, patch by me. ........ - -2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Whoops... - - * channels/chan_iax2.c: Let's make chan_iax2 media only native - transfers actually work. (issue #9376 reported by simone - cittadini) - - * channels/iax2-parser.c: Add TXMEDIA to list so that it is - properly displayed during iax2 packet output. - -2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Move the logic for destroying a call when no - response is received to a BYE outside of the block that checks - for FLAG_FATAL to be set. This flag is only set when the packet - is transmitted with the reliability set to XMIT_CRITICAL when the - original packet is transmitted. A BYE is always sent with it set - to XMIT_RELIABLE, meaning this code could never be encountered. - This resulted in seeing some SIP channels that would never go - away with the last packet sent being a BYE. (part of issue #9235, - patch from jcmoore) - -2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Contains a patch for fixing an encoding - problem when using Outlook to view voicemail emails and - attachments. This fix has also been tested on Thunderbird, - Evolution, Pine, and Mutt. (Issue 9336, reported by marwick, - patched by mutterc) - -2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Really ignore NULL frames and check whether - the channel hungup or not. (issue #9912 reported by junky) - - * /, main/app.c: Merged revisions 69127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 - lines Return group counting to previous behavior where you could - only have one group per category. (issue #9711 reported by - irroot) ........ - -2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Clarify a bit of logic. This doesn't change - behavior in any way, but it is helpful when following the logic - to debug problems like 9235. - - * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct - was accessed without the lock held. This issue was reported to me - via email by Dmitry Mishchenko. Thanks! - - * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois - in #asterisk-bugs. PQclear() was not called on the result - structure after doing a PQexec(). Also, fix up some formatting in - passing. - -2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Change the full frame dropping log message - to debug to avoid future bug reports. - - * channels/chan_iax2.c: Schedule the sending of a PING packet a - second later than previously so that it does not collide with the - LAGRQ. - -2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell@digium.com> - - * main/channel.c: In ast_channel_make_compatible(), just return if - the channels' read and write formats already match up. There are - code paths that call this function on a pair of channels multiple - times. This made calls fail that were using g729 in some cases. - The reason is that codec_g729a will unregister itself from the - list of available translators will all licenses are in use. So, - the first time the function got called, the right translation - path was allocated. However, the second time it got called, the - code would not find a translation path to/from g729 and make the - call fail, even if the channel actually already had a g729 - translation path allocated. (SPD-32) - -2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 68921 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 - lines Bring RTP back to Asterisk at the end of a native bridge no - matter what. ........ - -2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker@digium.com> - - * include/asterisk/time.h: Solaris 10 sometimes (?) needs this - include in order to have NULL defined. - -2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly - used - -2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 68732 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | - 1 line added check for NULL Pointer when calling misdn_new. - Asterisk does not allow us to create channels anymore when stop - gracefully is used :). also modified the restart_indicator to 0 - ........ - -2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 68682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 - lines Improve deadlock handling of the channel list. (issue #8376 - reported by one47) ........ - -2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | - 1 line fixed problem that the dummybc chanels had no lock, - checking for the lock now. Also fixed the channel restart stuff, - we can now specify and restart particular channels too. ........ - -2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_config.c: "dialplan save" produced garbage in the config - file - -2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell@digium.com> - - * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | - 4 lines Don't automatically hang up after running Dictate so that - callers can exit cleanly using '#' (closes issue #9577, patch - from Thomas Andrews) ........ - -2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: actually remember the type/subclass of full - frames that are in process - -2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp@digium.com> - - * /, main/say.c: Merged revisions 68397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 - lines Don't call ast_waitstream_full when the control file - descriptor and audio file descriptor are not set, simply call - ast_waitstream! (issue #8530 reported by rickead2000) ........ - - * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 - lines Do a DNS lookup immediately upon calling the dnsmgr - function, don't wait until a refresh happens. (issue #9097 - reported by plack) ........ - -2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell@digium.com> - - * /, main/say.c: Merged revisions 68351 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | - 3 lines Fix a problem where saying a character wouldn't properly - break out when the caller pressed '#' (issue #8113, reported by - patbaker82, patch from jamesgolovich (hey, long time no see!) and - patbaker82) ........ - -2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix incorrect French syntax of "old - messages". Request for feedback was sent to asterisk-dev mailing - list, with little response. Issue 9118, patch by junky. - -2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: some improvements to the IAX2 full frame - dropping logic recently added: - use inaddrcmp(), since we have - it - output the type of frame and subclass being dropped, and the - type/subclass that is already being processed (which caused the - drop) - -2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent - queue members when using realtime configuration for queues. Also, - remove an unneeded leading slash for the astdb family. (issue - #9911, patch by atis) - -2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an issue with newer phones which - require packets be padded out to the correct length. Issue 9887, - patch by DEA. - - * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 - lines Don't try to save voicemail greetings unless the user - presses '1' to accept/save. Issue 9904, patch by me. ........ - -2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a - check to make sure that greetings get stored properly. (Issue - 8016, reported by edhorton, patched by alamantia with - modification by me. Thanks to Jason Parker for the advice on - this). - -2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej@edvina.net> - - * channels/chan_features.c: Disable chan_features by default in - menuselect - -2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell@digium.com> - - * main/strcompat.c: Include stdarg.h for build issues on Solaris - (issue #9381) - -2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Fix logic when doing a name based channel search - for a structure when you want to start from a specific point in - the channel list. (issue #9324 reported by slavon) - - * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 - lines Allow the 'g' option to work if used with the 'S' option. - (issue #9888 reported by gasparz) ........ - -2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej@edvina.net> - - * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't - forget. - - * res/res_jabber.c: Ok, we found out that this is not about if you - have any *active* clients using TLS, but if you have initialized - TLS at all during the lifetime of the module. So if you reload to - disable TLS, it won't help. - - * res/res_jabber.c: If you have a jabber client that uses TLS, - refuse unload. Bad fix, but will prevent crashes while we are - trying to find a workaround. Iksemel development seems to have - stalled and we might have to stop using the TCP/TLS connections - in that library and use our own, which would scale better from a - poll/select perspective I guess. It would also make it easier to - migrate to OpenSSL and stop Asterisk from depending on both - OpenSSL and GnuTLS. - - * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make - sure we can unload res_jabber. Patch by phsultan - thanks! Due to - a bug in the iksemel library, this will not work if you are using - GTLS in the connection. That's being investigated. If you figure - out a way to handle that without us having to patch iksemel, let - us know in the bug report. Thanks. - -2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 - lines Only notify the devicestate system of a peer state change - when the peer is built from the config file. (issue #9900 - reported by arkadia) ........ - - * main/file.c: Properly handle cases where a stream can't be - written to. (issue #9757 reported by junky) - -2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell@digium.com> - - * res/res_snmp.c: Disable reload functionality in res_snmp. It is - not possible to initialize the snmp library more than once - without completely unloading the module and loading it again. - (issue #9571, reported by hristo, additional helpful debug - information from festr, patch from me) - - * channels/chan_sip.c: Fix a crash when doing call pickups with SIP - phones. The code unlocked the channel when it should not have. - (issue #9652, reported by corruptor, fixed by me) - -2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault - under a specific set of circumstances: 1. VoiceMailMain was - configured in the dialplan with an extension as its argument 2. A - message was left for this mailbox 3. Tried to call VoiceMailMain - but hung up before entering password. This was fixed by checking - that a pointer was non-null prior to trying to dereference it. - (Issue 9810, reported by xmarksthespot, patched by Corydon76 with - modifications by me). - -2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell@digium.com> - - * main/channel.c, /, include/asterisk/linkedlists.h: Merged - revisions 67715 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | - 5 lines We have some bug reports showing crashes due to a double - free of a channel. Add a sanity check to ast_channel_free() to - make sure we don't go on trying to free a channel that wasn't - found in the channel list. (issue #8850, and others...) ........ - -2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 67649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 - lines Reinvite the RTP back to the Asterisk machine when the - timeout happens. (issue #9888 reported by gasparz) ........ - - * main/translate.c: Fix plc_samples warning when registering a - translator. (issue #9897 reported by xylome) - - * apps/app_directed_pickup.c: Include macroexten while searching - for a channel to pick up in case they are in a macro. (issue - #9491 reported by jamesb63) - - * res/res_agi.c: Make the new "agi debug off" CLI command work. - (issue #9890 reported by eliel) - - * /, main/devicestate.c: Merged revisions 67593 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 - lines Revert channel name splitting fix for Zap. The moral of the - story is don't use - in your user/peer names. (issue #9668 - reported by stevedavies) ........ - -2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Fix some crashes related to the use of the - "meetme" CLI command. The code for this command was not locking - the conference list at all. (issue #9351, reported by and patch - submitted by Junk-Y, committed patch is different and by me) - -2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf@digium.com> - - * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug - 9883, wherein macros were not allowing the includes construct. - fixed and tested, looks OK. Now includes can serve as an adjunct - to catch. - -2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell@digium.com> - - * include/asterisk/linkedlists.h: This bug has been hanging over my - head ever since I wrote this SLA code. Every time I tried to go - debug it by adding some debug output, the behavior would change. - It turns out I wasn't crazy. I had the following piece of code: - if (remove) AST_LIST_REMOVE_CURRENT(...); Well, - AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my - conditional statement didn't do much good at all. It always ran - at least all of the macro minus the first statement, so I was - seeing list entries magically disappear when they weren't - supposed to. After many hours of debugging, I have come to this - extremely irritating fix. :) (issues #9581, #9497) - - * channels/chan_zap.c: Suppress a bunch of debug output unless - option_debug is on - -2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails - saved to IMAP storage using extensions other than gsm were unable - to be played over the phone. (Issue 9786, reporter: - xmarksthespot, Patched by xmarksthe spot with revisions by me, - reviewed by Russell Bryant). - -2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Correctly update date/time on devices - throughout the life of the device, instead of just at - registration. Issue 9152, yet another patch by DEA. - -2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: Added code to automatically add a default case to - switches that don't have one. In some cases, rather than fall - thru, it results in a goto with -1 result, which terminates the - extension; a sort of dialplan seqfault, sort of. This was - required to fix bug reported in 9881 - -2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell@digium.com> - - * main/channel.c: Handle a failure in malloc() in - ast_safe_string_alloc() - - * main/channel.c: Fix a problem that showed itself by causing Zap - channel names to be completely bogus on my machine. - ast_safe_string_alloc() was broken. It called vsnprintf() on a - va_args list twice without re-initializing it. After the first - usage, va_end() and va_start() must be called again. - -2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | - 1 line briding is a bool, fixed copy and paste issue. ........ - - * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 - Jun 2007) | 1 line simplified the EVENT_SETUP handling in the - cb_events function a lot. Commented the different possibilities a - bit and made functions of shared code. When the dialed extension - does not exist in the extensions.conf we'll jump into the 'i' - extension if this does exist, else we disconnect the call with - the cause:1 = No Route to Destination. ........ - -2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell@digium.com> - - * main/asterisk.c, main/loader.c, include/asterisk/module.h: When - shutting down "gracefully", go through and run the unload() - callbacks for all of the modules. "stop now" is considered a - non-graceful shutdown and will not go through this process. - (issue #9804, reported by chrisost, patch by me) - -2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Only muck with the thread structure if an - idle one was found/created. - -2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that a burst of full frames - (AST_FRAME_DTMF being the prime example) will not be processed - out of order... this is a brute force fix, but seems to be the - safest fix for now (thanks to the Digium PQ department for - finding this bug) - -2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h: Merged revisions 67209 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | - 1 line added possibility to deactivate bridging per port ........ - -2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007) - | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops. - ........ - -2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt - structure can disappear and the code did not account for it and - crashes. (issues #9642, #9569, #9666, probably others ... based - on the work by stevedavies and mihai, with additional changes - from me) - -2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix for skinny keepalives. If there is no - traffic from the phone for (keep_alive * 1100) ms (arbitrarily - adding 10% for network issues, etc), unregister the device. Issue - 8394, patch by DEA. - - * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar - to the recent fix for chan_skinny) Issue 9855, patch by DEA. - -2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Add comments for two functions that get - called with the appropriate call locked, but perform operations - that could result in the pvt structure getting destroyed before - returning again, causing numerous seg faults all over the module. - (inspired by issues #9642, #9569, and #9666, and the work done by - stevedavies and mihai) - -2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf@digium.com> - - * main/cdr.c: This typo has been here since 1.4 forked. It has been - the source of heartburn to many a dialplan/CDR programmer. - -2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell@digium.com> - - * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC) - -2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Better handle SIP devices that say they have - SDP content... but really don't. (issue #9398 reported by - mthomasslo) - - * apps/app_dial.c: Initialize cidname variable to nothing since it - may be used without having been touched. (issue #9661 reported by - dimas) - - * res/res_features.c: Returning a value that indicates the parking - of a call was a success when it really wasn't (because the - parking slot selected was in use) is the wrong thing to do. - (issue #9723 reported by mdu113) - -2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <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: Merged revisions 67060 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) - | 2 lines Add revision Id tags (by request of tzafrir) ........ - -2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Change the configure script to build a - test program against libcurl to make sure the results from - curl-config can be used to compile successfully. This is intended - to help prevent a situation where you are cross compiling, and - the configure script finds the curl library installed on the - host. (issue #9865, reported and patched by zandbelt) - -2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash - -2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When - handling an implicit ACK to a frame that was marked as the final - transmission for a call, don't call iax2_destroy() for that call - while the global frame queue is still locked. There is a very - nice explanation of the deadlock in the report. (issue #9663, - thorough report and patch from stevedavies, additional positive - test reports from mihai and joff_oconnell) - - * include/asterisk/stringfields.h: Fix some compiler warnings in - C++ modules. (issue #9866, reported by osk, patch by Corydon76) - -2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_odbc.c: On some drivers, deallocating the statement - handle isn't enough. We also have to clear the cursor (nice, - Oracle) - -2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removing extraneous debugging lines from - revision 66897. Sorry :) - - * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP - storage. Attachments with format specified as gsm were duplicated - (i.e. two attachments) were left. Thank you very much to - xmarksthespot for submitting the patch that fixed this. (Issues - 9787 and 8873, Reported by xmarksthespot and jerjer, patched by - xmarksthespot) - -2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell@digium.com> - - * channels/chan_skinny.c: Changes to the way DTMF is handled in the - core broke dialing in chan_skinny. This patch makes chan_skinny - usable again. I did not end up testing this, but there are - multiple positive test reports listed in the bug report. (issue - #9596, reported by pj, testing by pj and mvanbaak, and the fix - was written by DEA) - - * apps/app_page.c: List app_meetme as a module that app_page - depends on. - -2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * doc/asterisk.8: Issue 9850 - update preferred command line syntax - -2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell@digium.com> - - * res/res_speech.c, include/asterisk/app.h, - include/asterisk/speech.h: Change a couple of header files to not - use "new", which is a reserved keyword in C++. (issue #9830, - reported by osk) - -2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) - | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime. - Issue 8329 will remain unfixed for pbx_realtime, but only because - we lack core API to do it. ........ - -2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 - lines It is now possible for this path of execution to have the - frame pointer be NULL, therefore we need to check for it before - trying to access it. (issue #9836 reported by barthpbx) ........ - -2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixed seg-faults when recording greetings - in voicemail with IMAP enabled. (Issue No. 9735, reported by - xmarksthespot, patched by me) - -2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Silly me for having out of date source! Oh - well... I'm still leaving my comment. - - * channels/chan_sip.c: When calling some peer/host that may not - exist/reply back... don't keep the dialog in memory for all of - eternity. - - * channels/chan_zap.c, channels/chan_features.c: Change how channel - names are generated a bit. (issue #9825 reported by eldadran) - -2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) - | 2 lines If the value of a variable passed to FIELDQTY is blank, - then FIELDQTY should return 0, not 1. ........ - -2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Properly handle 408 request timeout - - according to the RFC, the dialog dies if a request in a dialog - gets this response. - - * channels/chan_sip.c: Don't issue hangup on hangup on hangup on - hangup (for jcmoore) - -2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Handle cases where a frame may have no data. (issue - #9519 reported by dmb) - -2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't reset hangupcause if we already have - one - - * channels/chan_sip.c: Tracking down hanging channels, killing them - one by one. Issue #9235 and related - -2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp@digium.com> - - * doc/datastores.txt: Update datastores documentation. (issue #9801 - reported by mnicholson) - -2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 - lines Issue #9802 - Change inuse counter on CANCEL ........ - -2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Make the usedistinctiveringdetection option - work again. (issue #9823 reported by premeau) - -2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: I don't know what this was trying to do, but - it's clearly incorrect. Issues 9808 and 9809. - -2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: have to check for OSP toolkit _after_ - checking for OpenSSL - -2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, main/say.c: Merged revisions 66127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) - | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch - ........ - -2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac, channels/chan_gtalk.c, makeopts.in, - res/res_jabber.c: handle the GNUTLS library properly in the - configure script and build system don't build in OSP support - unless we have found and are allowed to use SSL support - -2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell@digium.com> - - * main/channel.c: if the string field init fails, clean up the - stuff that was allocated already - -2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Fix slinfactory logic when dealing with - frames coming in that may already be in the signed linear format. - -2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell@digium.com> - - * main/channel.c: Check the result of ast_string_field_init() in - ast_channel_alloc() - - * main/rtp.c: Make 1.4 build on my machine, too.. - -2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker@digium.com> - - * configure: Rebuild configure script for previous ar fix. - - * configure.ac: Following moving strip to AC_PATH_TOOL, we need to - do something similar for ar. - -2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: - Checking for the strip application needs to be done with - AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross - compilation environments. - - * Makefile: Clear CFLAGS before running make for menuselect. (issue - #9784, reported by ovi, patch by me) - -2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_gtalk.c: oops, use #ifdef instead of #if - - * channels/chan_gtalk.c: don't reference GnuTLS headers and - functions unless the configure script found it - - * main/rtp.c: don't use uninitialized variables - -2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Add the ability to blacklist certain commands - from being executed using the Command AMI action. (issue #9240 - reported by junky) - -2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk - core dump since the GnuTLS interface did not support - multithreading correctly. - - * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN. - Patch by phsultan. Thanks! - -2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker@digium.com> - - * .cleancount: Update cleancount for that last commit - just for - good measure. - - * include/asterisk/translate.h, codecs/codec_speex.c, - main/translate.c, codecs/codec_ilbc.c: Fix handling of - zero-length frames when a codec is capable of native PLC. Issue - 9183, patch by Mihai. - -2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard@digium.com> - - * funcs/func_math.c: merged qwell's func_math patch for issue 9507 - -2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: I like it when the RTP stack compiles myself... - -2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues - when calling from gtalk to SIP over nat. - -2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell@digium.com> - - * apps/app_festival.c: Ensure that frames are fully initialized. - This will probably fix getting weird timestamp log messages in - logs when using the Festival app. (issue #9781, patch by me) - - * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous - code would result in oscillating and incorrect data. - Additionally, the RTT would sometimes report negative values due - to incorrect calculations. (issue #9601, patch from davetroy) - -2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for - jingle - -2007-05-24 14:42 +0000 [r65839] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 - lines Allow RFC2833 to be negotiated when an INVITE comes in - without SDP and is not matched to a user or peer. (issue #9546 - reported by mcrawford) ........ - -2007-05-24 14:38 +0000 [r65836] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan - - Fix "login" as component to jabber server. ...and, by accident, - fix a bug in chan_sip for stopping a loop on retransmits of BYE - requests. - -2007-05-24 09:37 +0000 [r65768] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 - Mai 2007) | 1 line we should only activate the generator in - chan_misdn, when asterisk hask not yet taken the call - (WAITING4DIGS state). Alerting audio will be generated fomr - asterisk for example. ........ - -2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: start the delayed PBX when receive voice or - video full frames as well, and comment this delayed-PBX activity - - * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) - | 2 lines ensure that variables are set on a newly created - channel before we start a PBX on it ........ - - * channels/chan_iax2.c: clear the 'delay PBX' flag when we are - ready to start the PBX - - * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2 - channel until we have some sort of response to our ACCEPT (ACK or - anything else) - - * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) - | 2 lines if we are going to set variables on a newly created - channel, it should be done *before* we start the PBX on it - ........ - -2007-05-23 13:07 +0000 [r65589] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | - 3 lines Revert revision 62417 as someone reported problems with - it to Mark. This was related to issue #9588. ........ - -2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_version: when building a version string for a - developer branch, include the base branch in the version string - -2007-05-22 18:40 +0000 [r65501] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a - dependency for app_voicemail and chan_zap (Thanks to mnicholson - for pointing it out) - -2007-05-22 15:04 +0000 [r65452] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Remove a double const. - -2007-05-22 14:02 +0000 [r65408] BJ Weschke <bweschke@btwtech.com> - - * apps/app_followme.c: Fix a problem with flag recognition. - -2007-05-22 13:09 +0000 [r65394] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | - 4 lines Fix a memory leak that I just noticed in the device state - handling in app_queue. On most device state changes, it would - leak roughly 8 to 64 bytes (the length of the name of the - device). ........ - -2007-05-22 08:12 +0000 [r65342] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 - Mai 2007) | 1 line we stop the tones only when we're in the - pre-call phase, otherwise e.g. when in CONNECTED state we should - not stop tones when we receive an Information Message ........ - -2007-05-20 17:59 +0000 [r65250] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: res_agi needs to export two symbols - (ast_agi_register and ast_agi_unregister) for usage by others. - (issue #9755 reported by mnicholson) - -2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy <murf@digium.com> - - * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c - to main/cdr.c, and neither did I. This is the remainder of the - 9717 patch, the fix for the run-away FAIL status for a call - - * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions - 65172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 - line This update will fix the situation that occurs as described - by 9717, where when several targets are specified for a dial, if - any one them reports FAIL, the whole call gets FAIL, even though - others were ringing OK. I rearranged the priorities, so that a - new disposition, NULL, is at the lowest level, and the - disposition get init'd to NULL. Then, next up is FAIL, and next - up is BUSY, then NOANSWER, then ANSWERED. All the related set - routines will only do so if the disposition value to be set to is - greater than what's already there. This gives the intended - effect. So, if all the targets are busy, you'd get BUSY for the - call disposition. If all get BUSY, but one, and that one rings is - not answered, you get NOANSWER. If by some freak of nature, the - NULL value doesn't get overridden, then the disp2str routine will - report NOANSWER as before. ........ - -2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 - lines Not getting an ACK to a 200 OK in the initial invite is - critical to the call. ........ - - * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 - lines Issue 9235 - part of the problem, maybe not all. Please - retry with this patch (and no other patch) if you have problems - with hanging SIP channels. Thank you. A special Thank You to - WeBRainstorm that gave me access to his system. ........ - - * channels/chan_sip.c: - Adding support for putting calls OFF hold - with a re-invite with blank SDP. This was a bug found while doing - tests at SIPit in Antwerp. - In order to not duplicate code, I - restructured some of the code for putting calls on/off hold. - Thanks DEA for reminding me. This fix has been asleep in the - videocaps branch until now. - -2007-05-18 12:40 +0000 [r65039] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged - revisions 65007 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | - 1 line fixed a warning regarding Keypad encoding. encode the IE - sending_complete at the right position. ........ - -2007-05-18 10:37 +0000 [r64974] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of - call - -2007-05-18 08:58 +0000 [r64904] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 - Mai 2007) | 1 line we *need* to send a PROCEEDING when - sending_complete is set, even if need_more_infos is requested. - ........ - -2007-05-18 02:48 +0000 [r64868] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix a small bug I noticed while working on - something else. app_queue did not unregister its device state - monitoring callback in unload_module(). So, this would make - Asterisk crash on the first device state change after you unload - the module. - -2007-05-17 21:19 +0000 [r64820] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 64819 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) - | 2 lines How is it that we never caught that this is returning - the opposite of our documentation, until now? ........ - -2007-05-17 16:53 +0000 [r64761] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 - lines If we have a negative current message, we shouldn't go back - even further... Issue 9727. ........ - -2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant <russell@digium.com> - - * contrib/scripts/astxs (removed): Remove script that is no longer - functional since the build system was redone. (issue #9340, - reported by junky) - - * apps/app_dial.c: Increase the size of a buffer to support longer - dial strings for channels. (issue #9291, reported and fix - suggested by meni) - -2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Even more direct RTP setup fixes! Don't - allow a codec that isn't supported to creep into the SDP of - either side. (issue #9446 reported by marcelbarbulescu) - - * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported - by xmarksthespot) - -2007-05-17 06:13 +0000 [r64686] Russell Bryant <russell@digium.com> - - * README: Update the main README to reflect the new build process - for 1.4 and above. (issue #9725, patch by eliel) - -2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson <oej@edvina.net> - - * /: Blocking patch already in this code - - * channels/chan_sip.c: Fix auth on BYE. (Different patch than for - 1.2) - - * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE - - * channels/chan_sip.c: Final part of issue #9483 - fixing - transfer() of sip calls in the dial plan (twilson) - - * channels/chan_sip.c: Issue #9439 - properly handle username - parameters in SIP uri. - - * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 - lines Support SIP uri's starting with SIP: and sip: (reported by - Tony Mountfield on the mailing list. Thanks!) ........ - - * /, channels/chan_sip.c: Merged following patch with a lot of - changes for 1.4 ------ Merged revisions 64514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 - lines Issue #9726 - rlister - Better logging for ACL denials - While at it, also added better logging and handling of peers that - are not supposed to register. My patch, stole the issue report - from Russell. My apologies, Russell :-) ........ - -2007-05-16 08:44 +0000 [r64515] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 - Mai 2007) | 1 line in the case immediate=yes, we directly jump - into the dialplan, where people can use PlayTones to indicate a - Dialtone, so we don't need to to that by ourself. also we should - not do a dialtone_indicate for incoming calls on a TE port in - overlapdialmode. ........ - -2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant <russell@digium.com> - - * res/res_features.c: Properly fix a problem that occurs when you - set PARKINGEXTEN to an exten where a call is already parked. - (issue #9723, patch by me) - - * res/res_features.c: When someone requests a specific parking - space using the PARKINGEXTEN variable, ensure that no other - caller is already there. (issue #9723, reported by mdu113, patch - by me) - -2007-05-14 19:26 +0000 [r64324] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit - more - -2007-05-14 19:13 +0000 [r64306] Russell Bryant <russell@digium.com> - - * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by - just ignoring it. An unknown indication will trigger an error and - cause sounds to stop, which in this case, is ringing. - -2007-05-14 18:52 +0000 [r64280] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Handle network errors, like host or network - unreachable, in a better way. This means that calls to hosts or - qualify (OPTION) messages will fail quicker if the TCP/IP stack - tells us that there is an issue. Since this is an unconnected UDP - socket, we will not get error messages directly in most cases, - but maybe on the second and third try. This is already - implemented in trunk. - -2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp <jcolp@digium.com> - - * codecs/codec_speex.c: Properly set datalen field when doing PLC - in codec_speex. (issue #9722 reported by mihai) - - * /, main/devicestate.c: Merged revisions 64275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 - lines Only perform stripping of - strings from the channel name - for Zap channels. Anywhere else we might remove a legitimate part - of a device name. (issue #9668 reported by stevedavies) ........ - - * main/channel.c: Fix scenario where if a phone that simply called - Echo() put itself on hold it could never get off hold. - -2007-05-14 13:58 +0000 [r64193] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570, - worrisome CDR warnings have been removed, that are either not - helpful, or not relevant. - -2007-05-14 10:39 +0000 [r64157] Olle Johansson <oej@edvina.net> - - * main/channel.c: Add hangupcause when we lack codecs for - transcoding - -2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: This concludes my final adventure with - bitmasks and the onhold flag. Would anyone care for some peanuts? - - * channels/chan_sip.c: Tweak hold flags some more. They can be of - three states when active: active, inactive, one direction. - - * channels/chan_sip.c: Ensure the onhold flag is set no matter what - when being put on hold. - -2007-05-11 20:16 +0000 [r63982] Jason Parker <jparker@digium.com> - - * main/manager.c: Hide manager password from "manager show user - foo". I realize that there are other ways to get this, but we - really don't need to just show it in plain text so easily. Issue - 9273, patch by junky - -2007-05-11 16:35 +0000 [r63905] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions - 63903 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) - | 2 lines Issue 9121 - fixups for safe_asterisk script ........ - -2007-05-11 16:05 +0000 [r63886] Russell Bryant <russell@digium.com> - - * main/manager.c: When MD5 authentication is not possible because - there is no challenge present, either because the Challenge - action was never issued, or some other reason, give a proper - error message and return an error instead of claiming that the - user wasn't found. (reported by jsmith on IRC) - -2007-05-11 15:43 +0000 [r63872] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Make the PARKINGEXTEN feature of parking - actually work. (issue #9708 reported by mdu113) - -2007-05-10 23:15 +0000 [r63830] Jason Parker <jparker@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 - lines Fix an issue with trying to kill a thread before it gets - created. Issue 9709, patch by nic_bellamy. ........ - -2007-05-10 22:23 +0000 [r63804] Russell Bryant <russell@digium.com> - - * main/manager.c: Strip terminal escape sequences from CLI command - output that is going to be sent out over the manager interface. - (issue #9659, reported by pari, fixed by me) - -2007-05-10 20:48 +0000 [r63750] Doug Bailey <dbailey@digium.com> - - * main/callerid.c: Add test for negative offsets in cid data to - prevent infinite loops. - -2007-05-10 20:46 +0000 [r63749] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 - lines Do not allocate SIP pvt's for PEERs we can not reach. This - was seen as a lot of dialogs being created then immediately - destroyed at reload/restart of the SIP channel. ........ - -2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Use the DTMF frame on the channel when returning - a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. - - * channels/chan_sip.c: Do not prematurely go on hold if sendonly - was not actually set. - -2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 - lines Make sure we only create a DSP if it's requested on - SUB_REAL ........ - -2007-05-09 16:55 +0000 [r63612] Russell Bryant <russell@digium.com> - - * main/channel.c: Modify ast_senddigit_begin() to use the same - assumptions used elsewhere in the code in that if a channel does - not have a send_digit_begin() callback, it only cares about DTMF - END events. (pointed out by Michael Neuhauser on the asterisk-dev - list) - -2007-05-09 16:54 +0000 [r63611] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 - lines Properly handle hints that point to multiple devices in - chan_sip. Why chan_sip is even doing this I have no idea but I - would rather not go into a rant. (issue #9536 reported by - rlister) ........ - -2007-05-09 16:43 +0000 [r63608] Russell Bryant <russell@digium.com> - - * main/channel.c: Only call ast_senddigit_begin() in - ast_senddigit() if the channel has a send_digit_begin() callback. - Checking the END_DTMF_ONLY flag was the wrong thing to do, - because that flag indicates that a *bridged* channel only wants - DTMF END events coming from this channel. - -2007-05-09 14:50 +0000 [r63566] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) - | 2 lines Replicate fix from 51158 (app_voicemail) to - app_directory (Issue 9224) ........ - -2007-05-09 13:24 +0000 [r63535] Russell Bryant <russell@digium.com> - - * Makefile: I have seen multiple people post questions trying to - figure out what the message "The configure script must be - executed before running 'make'" means. So, add another like that - says to specifically run ./configure. If this isn't obvious - enough, then they should be using something like AsteriskNOW and - not installing from source. - -2007-05-09 13:17 +0000 [r63534] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, - channels/misdn/isdn_msg_parser.c: Merged revisions - 62945,63402,63519 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | - 1 line when we're in state WAITING4DIGS, we use the asterisk - tone-generator which prods us, so we can't just return -1 in - misdn_write in this case. Added a MISDN_KEYPAD channel variable, - and fixed the sending of keypad. this enables us to modify the - call forward parameters in the switch. ........ r63402 | crichter - | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added - application misdn_check_l2l1 which tries to pull up the L1/L2 on - all ports that have the layers down in a group. It waits then for - a timeout. This helps for scenarios where multiple PMP BRIs are - grouped together, or where a provider has a faulty PTP - Implementation, that looses the L2 after a while. ........ r63519 - | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line - release_chan frees ch, so we should never touch ch after - release_chan, this may cause segfaults. ........ - -2007-05-09 13:04 +0000 [r63532] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status. - (Reported on asterisk-users) - -2007-05-08 22:38 +0000 [r63478] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) - | 2 lines Issue 9602 - segfault in app_macro ........ - -2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant <russell@digium.com> - - * res/res_features.c: I mixed up the use of the find_feature() - function, so I renamed it find_dynamic_feature, and changed the - code to use the correct lock when using it. - - * res/res_features.c: Use a read/write lock when accessing the - built-in features. - - * contrib/scripts/realtime_pgsql.sql (added), - contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to - contrib/scripts to be with the rest of the sql examples. (issue - #9676, suretec) - -2007-05-08 06:22 +0000 [r63360] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) - | 2 lines Issue 9527 - upon entering a folder, no message is - selected (curmsg == -1), so deleting causes memory corruption - (beyond bounds) ........ - -2007-05-07 22:28 +0000 [r63329] Russell Bryant <russell@digium.com> - - * configs/res_pgsql.conf.sample (added), - configs/extconfig.conf.sample, contrib/realtime_pgsql.sql - (added): Add a sample configuration file and example tables for - use with res_config_pgsql. (issue #9676, suretec) - -2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp <jcolp@digium.com> - - * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged - revisions 63285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 - lines Properly handle what happens during a masquerade in - relation to group counting. (issue #9657 reported by ramonpeek) - ........ - - * channels/chan_sip.c: Minor backport of revision 59083 in trunk. - Don't queue an unhold frame up if the call was never on hold to - begin with. - -2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson <oej@edvina.net> - - * main/config.c: Don't remove configuration from memory just - because one section failed. - - * /: Guess svnmerge doesn't handle files that move around. Blocking - patch to ./config.c - -2007-05-06 12:28 +0000 [r63152] Olle Johansson <oej@edvina.net> - - * main/file.c: Stop the video stream when you stop playback of all - streams for a call - -2007-05-04 20:03 +0000 [r63099] Jason Parker <jparker@digium.com> - - * res/res_jabber.c: Fix a crash when checking version attribute in - an incoming XML caps element. Issue 9667, patch by phsultan. - -2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni <paripurnachand@digium.com> - - * configs/manager.conf.sample: explanation for httptimeout in - manager.conf - -2007-05-03 16:44 +0000 [r62989] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 - lines When a peer is seeded or built tell the devicestate core to - update it's status. This is easier then having chan_sip load - before pbx_config. (issue #9658 reported by dlynes) ........ - -2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming <kpfleming@digium.com> - - * main/loader.c: improve loader a bit, by avoiding trying to - initialize embedded modules twice and avoiding trying to load - modules from disk when they have been loaded already during the - 'preload' pass (reported by blitzrage on IRC, patch by me) - -2007-05-03 15:23 +0000 [r62942] Russell Bryant <russell@digium.com> - - * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell - Bryant, 2007, TM, Patent Pending). This set of changes came from - a debugging session I had with Dwayne Hubbard. When he called - into his home FXO, ran the Echo application, and pressed a digit, - the digit would be echoed back and would never end. This is - fixed, along with a couple other little improvements. * When - chan_zap is in the middle of playing a digit to a channel, it - feeds back null frames, not voice frames. So, I have modified - ast_read to check the timing on emulated DTMF when it receives - null frames, in addition to where it was doing this on voice - frames. * Make a tweak to setting the duration on emulated DTMF - digits. If there was no duration specified, it set it to be the - minimum, instead of the default. * Instead of timing the emulated - digits off of the number of samples in audio frames that pass - through, just use time values. Now there is no code in this - section that assumes 8kHz audio. - -2007-05-03 14:41 +0000 [r62913] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19 - (added), pbx/ael/ael-test/ael-test18/extensions.ael, - pbx/ael/ael-test/ael-test19/extensions.ael (added), - pbx/ael/ael-test/ael-test19 (added), - pbx/ael/ael-test/ref.ael-test20 (added), - pbx/ael/ael-test/ael-test20/extensions.ael (added), - pbx/ael/ael-test/ael-test20 (added): updated the ael regressions - to match what's in trunk - -2007-05-03 14:36 +0000 [r62912] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged - revisions 61357,61770,62885 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | - 1 line some fixes for PMP Hold/Retrieve, it should work now, when - briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 - (Di, 24 Apr 2007) | 1 line added lock for sending messages to - avoid double sending. shuffled some empty_chans after the - cb_event calls, this avoids that a release_complete from a quite - different call releases a fresh created setup by accident. - ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 - Mai 2007) | 1 line fixed the problem that misdn_write did not - return -1 when called with 0 samples in a frame this resultet in - a deadlock in some circumstances, when the call ended because of - a busy extension. added encoding of keypad. ........ - -2007-05-03 13:54 +0000 [r62883] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test18 (added), - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ael-test18/extensions.ael (added), - pbx/ael/ael-test/ael-test18 (added), - pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c, - pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7: - These mods fix bug 9623, where an '@' in the eswitch contents - causes a syntax error. I also updated the regressions. - -2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 - May 2007) | 2 lines doh... initializing the pointer variable will - work just a bit better ........ - - * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 - May 2007) | 7 lines increase reliability and efficiency of static - Realtime config loading via ODBC: don't request fields we aren't - going to use don't request sorting on fields that are pointless - to sort on explicitly request the fields we want, because we - can't expect the database to always return them in the order they - were created (reported by blitzrage in person (!), patch by me) - ........ - - * res/res_config_pgsql.c: improve static Realtime config loading - from PostgreSQL: don't request sorting on fields that are - pointless to sort on use ast_build_string() instead of snprintf() - don't request the list of fieldnames that resulted from the query - when we both knew what they were before we ran the query _AND_ we - aren't going to do anything with them anyway (patch by me, - inspired by blitzrage's bug report about res_config_odbc) - -2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant <russell@digium.com> - - * main/channel.c: Merge changes from team/russell/inband_dtmf ... - Fix some issues related to generating inband DTMF. There are two - changes here: 1) The list of DTMF tones in the senddigit_begin() - function explicitly specified 100ms of the tone followed by 100ms - of silence. This really broke things with the way that Asterisk - now wants complete control over when the digit begins and ends. - So, regardless of what Asterisk really wanted to do, this was - going to play out the tone at the length it wanted to. This - caused various problems like DTMF translation to inband to be - extremely unreliable. The list of tones has been changed so that - the correct DTMF tone is played indefinitely until Asterisk tells - it to stop. 2) ast_write() had to be modified to let a DTMF_END - frame get processed even when a generator is present. This is how - the tone will finally get stopped. (issues #8944, #9250, #9348, - maybe others. Thanks to mdu113 from #8944 for the testing and - feedback!) - - * main/manager.c: Backport the change that only went in to trunk - that fixes the command manager action over http. (reported - internally by pari and bkruse) - -2007-05-02 20:46 +0000 [r62738] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May - 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being - in 'h' extension louses up the dst field ........ - -2007-05-02 17:43 +0000 [r62692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) - | 4 lines Issue 9638 - if a text frame is sent with no - terminating NULL through a bridged IAX connection, the remote end - will receive garbage characters tacked onto the end. ........ - -2007-05-02 17:10 +0000 [r62689] Steve Murphy <murf@digium.com> - - * configs/extensions.conf.sample, main/channel.c, main/pbx.c, - channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the - clid, src fields in channel_alloc call. b)in the channel_alloc - func, set the cid_num and name fields from the arglist[blush]. c) - don't update the channel app & app data fields if you are in the - 'h' extension. d)the load_module func in cdr_radius needs to - return DECLINE, SUCCESS. - -2007-05-02 06:15 +0000 [r62624] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't unlock a channel that we already know - does not exist (propably isue 8228) - -2007-05-01 21:57 +0000 [r62548] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 62547 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | - 4 lines Remove an unnecessary check that makes it so if you hang - up after doing an attended transfer before the target extension - answers the channel, the transfer is not successful. (issue - #9338, patch by svanlund) ........ - -2007-05-01 21:34 +0000 [r62545] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user() - (found by rayjay, different fixes by me) - -2007-05-01 16:26 +0000 [r62497] Russell Bryant <russell@digium.com> - - * /, configs/indications.conf.sample: Merged revisions 62496 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | - 3 lines Add indications.conf information for the Philippines. - (issue #9525, reported and patched by loloski) ........ - -2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | - 4 lines This patch fixes an issue where depending on the cause - code, when the network sends a PRI disconnect, the call may not - be properly hung up. (issue #9588, reported and patched by - softins) ........ - - * include/asterisk/http.h, main/http.c: When serving dynamic - content, include a Cache-Control header to instruct the browsers - to not store the resulting content. (issue #9621, reported by - Pari, patch by me) - -2007-04-30 14:52 +0000 [r62371] Jason Parker <jparker@digium.com> - - * configs/iax.conf.sample: Remove unused (and potentially - confusing) jitterbuffer options from sample config. - -2007-04-30 14:36 +0000 [r62369] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c, /: Merged revisions 62368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 - lines Update copyright notice. It's now the year 2007! ........ - -2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix a bug that made the "language" setting - in zapata.conf not functional. (issue #9626, reported and fixed - by sergee) - - * apps/app_meetme.c: Note that the "talker optimization" option - will be enabled by default in 1.6 - -2007-04-27 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.4 released. - -2007-04-27 21:10 +0000 [r62218] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: Fix a weird problem where when a caller - talking to someone sitting behind an agent channel sent a digit, - the digit would be played to the agent for forever. This is - because chan_agent always returned -1 from its send_digit_begin - and _end callbacks. This non-zero return value indicates to the - Asterisk core that it would like an inband DTMF generator put on - the channel. However, this is the wrong thing to do. It should - *always* return 0, instead. When the digit begin and end - functions are called on the proxied channel, the underlying - channel will indicate whether inband DTMF is needed or not, and - the generator will be put on that one, and not the Agent channel. - (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed - by me) - -2007-04-27 16:17 +0000 [r62174] Jason Parker <jparker@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 - lines This transcoder message needn't be a NOTICE. I've seen it - cause confusion more than a few times. ........ - -2007-04-27 16:14 +0000 [r62171] Russell Bryant <russell@digium.com> - - * main/pbx.c: If no variables were passed into - pbx_substitute_variables_helper_full(), then don't even bother - creating a temporary bogus channel, since that is only for - allowing certain functions to operate on the variables as if they - were on a channel. Most importantly, this fixes a crash. (issue - #9613, reported by callguy, fixed by me) - -2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 - lines Issue #7351 - SIP Cancel fails due to the wrong contact - uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka - - THANKS!!!! THis was a hard one to catch. ........ - - * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some - annoying DEBUG messages not controlled by option_debug (DEA). - Thanks! - -2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 - lines Revert previous fix for when the IAX2 channel goes funky - (that's the technical term). This is causing legit calls to be - prematurely hung up. (issue #9600 reported by justdave) ........ - - * main/channel.c: Missed an ast_app_group_discard during merge. - Thanks blitzrage! - - * res/res_monitor.c: Don't always say that the channel is being - paused if it is actually being unpaused in the Manager ack - message. (reported by jsmith in #asterisk-bugs) - - * main/config.c, /: Merged revisions 61958 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 - lines Don't count failed include attempts against the - configuration include level. (issue #9593 reported by mostyn) - ........ - -2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) - | 2 lines handle a very bizarre race condition with channels - being redirected before a simple switch can be started on them - (issue #9286) ........ - -2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | - 2 lines If the callerid= option is specified, but empty, clear - any previous data. ........ - - * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | - 2 lines Ensure that callerid settings are reset on a reload. - ........ - -2007-04-25 19:21 +0000 [r61805] Joshua Colp <jcolp@digium.com> - - * main/cli.c, main/channel.c, include/asterisk/app.h, - funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 - lines Merge rewritten group counting support. No more storing - data on the variable list of the channels. That was bad, mmmk? - (issue #7497 reported by sabbathbh) ........ - -2007-04-25 16:22 +0000 [r61799] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | - 3 lines Fix a typo where cid_num got copied instead of cid_ani. - (issue #9587, reported and patched by xrg) ........ - -2007-04-24 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.3 released. - -2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 61786 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | - 4 lines Don't crash if a manager connection provides a username - that exists in manager.conf but does not have a password, and - also requests MD5 authentication. (ASA-2007-012) ........ - - * main/channel.c, include/asterisk/channel.h: Improve DTMF handling - in ast_read() even more in response to a discussion on the - asterisk-dev mailing list. I changed the enforced minimum length - of a digit from 100ms to 80ms. Furthermore, I made it now enforce - a gap of 45ms in between digits. These values are not - configurable in a configuration file right now, but they can be - easily changed near the top of main/channel.c. - -2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) - | 1 line removed #if 0 block from chan_phone, chan_zap, and - chan_modem restart_monitor() ........ - -2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell@digium.com> - - * main/dial.c: Add a few more state changes in - handle_frame_ownerless() so that the SLA code will get notified - of these changes even when an owner channel is not provided. This - isn't from a specific bug report, it's just something I noticed - while poking around. - -2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 - lines Allow RFC2833 to be sent in the response SDP when an INVITE - comes in without SDP. (issue #9546 reported by mcrawford) - ........ - -2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell@digium.com> - - * main/pbx.c: Some dialplan functions, such as CUT(), expect to - operate on variables on a channel. So, this little hack lets them - work in places where a channel doesn't exist, such as within - DUNDi configuration. (issue #9465, reported and patched by - Corydon76, testing by blitzrage) - - * main/channel.c: Ensure that digits passing through Asterisk have - a reasonable minimum length. It is currently 100 ms. If someone - thinks this should be different, feel free to speak up. (related - to issues #8944, #9250, and #9348) - -2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker@digium.com> - - * main/rtp.c: Avoid invalid seqno cycling detection. Per comment - from Dave Troy: This adds back in some simple typecasting I had - in an earlier version which I realize now may be breaking things. - Issue #9554. - - * main/loader.c, /: Merged revisions 61704 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 - lines Fix an issue that I noticed while looking over issue 9571. - The reload timestamp was getting set after reloading the built-in - stuff, and before the modules. ........ - -2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell@digium.com> - - * main/rtp.c: Remove a stray debug message introduced by a recent - commit. - -2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 - lines If the '* to hangup' option is not enabled, we don't need - to disable * as a valid exit key. If it was enabled, this - statement would've never been checked in the first place. Issue - #9552 ........ - -2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell@digium.com> - - * main/config.c, apps/app_voicemail.c, main/manager.c, - include/asterisk/config.h: Fix the UpdateConfig manager action to - properly treat "variables" and "objects" differently (a=b versus - a=>b). (issue #9568, reported by pari, patch by me) - -2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 - lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by - Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........ - -2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/manager.c: Bug 9557 - simple reason why reading a function - always returned NULL - - * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c, - funcs/func_groupcount.c, /, funcs/func_timeout.c, - funcs/func_cdr.c: Merged revisions 61680 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) - | 5 lines Bug 9557 - Specifying the GetVar AMI action without a - Channel parameter can cause Asterisk to crash. The reason this - needs to be fixed in the functions instead of in AMI is because - Channel can legitimately be NULL, such as when retrieving global - variables. ........ - -2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: allow external build systems to extract the - required sound file versions - -2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej@edvina.net> - - * main/rtp.c: Clean upp formatting, add some doxygen stuff while - we're in cleaning mode... Thanks Kevin! - - * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy) - -2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: #9483, half of patch by twilson to solve 302 - redirect issues - - * /: Blocking AstHoloPatch from 1.2 - -2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf@digium.com> - - * main/cdr.c: This is a fix to the way CDR merge handles the data - that results from ForkCDR. - -2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 - lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves - the same as OUTBOUND_GROUP except it will get unset after use so - it won't get accidentally inherited. (issue #BE-140) ........ - - * apps/app_speech_utils.c: Do not bother looking for a result if - none are present. - - * channels/chan_sip.c: For those very verbose SIP implementations - that attach tons of info to the Contact header... let's increase - our variable sizes. (issue #9535 reported by jeffg) - -2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Eliminate a compiler warning with - ODBC_STORAGE enabled so that it will build under dev-mode. - -2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf@digium.com> - - * channels/chan_oss.c: A fix for chan_oss that resulted from the - CDR changes; it helps to use the right info. - -2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't assume the callid of a dialog will be - set, as in some circumstances it may not. (issue #9534 reported - by tecnoxarxa) - -2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | - 5 lines If someone sets the "useragent" option in sip.conf to be - empty, then don't add the User-Agent header at all. It is an - optional header, anyway. Also, the bug report says that some of - Japan's SIP providers don't allow it for some weird reason. - (issue #9488, reported by makoto, fixed by me) ........ - -2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns@beronet.com> - - * channels/chan_misdn.c: Don't export AOCD variables on - misdn_hangup anymore, this was mainly a fix for trunk.. - -2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | - 6 lines Fix a bug with switching between host=dynamic and using - specific hosts for peers. The code would only reset the peer's - address when it is dynamic if it was a new peer structure. Now, - it will also reset the address if it was already in the peer - list, but before the reload, it was not dynamic. (issue #9515, - reported by caio1982, fixed by me) ........ - - * main/http.c: Add "svgz" to the mimetypes table. (issue #9510, - bkruse) In passing, constify the elements of the mimetypes table. - - * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | - 5 lines Remove the attempt at reporting configuration errors in - sip.conf. This can cause a bunch of improper messages when using - realtime. I give up. As oej tried to convince me when I put this - in, there is just no easy way to do it. (inspired by a message on - the -dev list) ........ - -2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns@beronet.com> - - * channels/chan_misdn.c: Export AOCD variables on misdn_hangup. - - * channels/chan_misdn.c: Ignore facility messages in case we don't - have a corresponding channel object. - - * channels/chan_misdn.c: AOCD's are now exported to asterisk - channel variables. - -2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell@digium.com> - - * main/Makefile, main/http.c, main/minimime (removed): File upload - support was added to solve some needs for the Asterisk GUI. - However, after much discussion, it has been decided that adding - this to 1.4 is not in the best interests of the project. It has - been removed here, but will remain in trunk. - -2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr - 2007) | 2 lines msns config parameter defaults to '*' ........ - -2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf@digium.com> - - * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a - previous fix to overcome a compiler warning; the app NoCDR() has - been updated to mark the channel CDR as POST_DISABLED instead of - destroying the CDR; this way its flags are propagated thru a - bridge and the CDR is actually dropped. The cases where only one - channel in a bridge has a CDR was cleaned up. - -2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 - lines - Don't send ActionID before Response: header. - Don't use - a blank in an AMI header ........ - -2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming@digium.com> - - * main/minimime/mm_envelope.c, res/res_features.c: fix up some - warnings found using --enable-dev-mode - - * main/minimime/Doxyfile (removed), - main/minimime/tests/messages/CVS (removed), - main/minimime/tests/CVS (removed): remove some more stuff we - don't need - -2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell@digium.com> - - * main/minimime/test (removed): Remove another directory that - should no longer be there - - * main/minimime/Make.conf (removed), main/minimime/mytest_files - (removed), main/minimime/.cvsignore (removed), main/minimime/sys - (removed), main/minimime/mm-docs (removed): Remove various files - that I thought I already removed. - -2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Use the appropriate interface name with - COMPLETECALLER. Issue 9395. - -2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf@digium.com> - - * channels/chan_oss.c, main/channel.c, main/cdr.c, - channels/chan_phone.c, channels/chan_misdn.c, - channels/chan_skinny.c, channels/chan_features.c, - channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, - channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, - channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, - channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, - include/asterisk/channel.h, channels/chan_gtalk.c, - channels/chan_iax2.c: This is a big improvement over the current - CDR fixes. It may still need refinement, but this won't have as - many folks bothered. - -2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej@edvina.net> - - * res/res_jabber.c: Add final new line after JabberEvent - -2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker@digium.com> - - * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 - lines Allow matching on names shorter than 3 chars. This also - fixes the case where somebody wants to match on less then 3 - chars. Issue 9071 ........ - -2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) - | 2 lines Don't check for error when lowering priority (according - to the manpage, it should never happen anyway). It might could - happen, though, if another thread messed with the priority, so - safeguard against that (reported via -dev list). ........ - - * channels/chan_local.c, /: Merged revisions 60846 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 - Apr 2007) | 2 lines Bug 9505 - If the return value for - local_queue_frame is set, then p->lock is no longer valid. - ........ - -2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 - lines When calling a device that then forwards us elsewhere... we - have to make our channels compatible if it is the only channel - being dialed. (issue #9445 reported by marcelbarbulescu) ........ - - * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if - MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy) - -2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007) - | 2 lines Gosub called within a Macro resets the arguments - improperly and causes general weirdness. (Issue 8329) ........ - - * main/http.c: Fix --enable-dev-mode - - * channels/chan_oss.c: Off by one error, resulting in a crash - (Issue 9500) - - * /, main/file.c: Merged revisions 60660 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) - | 2 lines Bug 9486 - memory leak when opening a filestream - ........ - -2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell@digium.com> - - * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile, - main/minimime/mimeparser.yy.c, main/minimime/minimime.c, - main/manager.c, main/minimime/mm_mimepart.c, - main/minimime/test.sh, configure, include/asterisk/compat.h, - main/strcompat.c, main/minimime/mm_internal.h, main/http.c, - main/minimime/tests/parse.c, main/minimime/mm_base64.c, - main/minimime/mm_mimeutil.c, main/minimime/mm.h, - main/minimime/tests, main/minimime/mm_header.c, - main/minimime/mm_error.c, main/Makefile, - main/minimime/mm_codecs.c, main/minimime/mm_param.c, - configure.ac, main/minimime/Makefile, main/minimime/mm_init.c, - include/asterisk/manager.h, main/minimime/strlcpy.c, - configs/http.conf.sample, main/minimime/mm_parse.c, - main/minimime/tests/create.c, main/minimime/mm_contenttype.c, - main/minimime/mm_util.c, main/minimime/mm_envelope.c, - main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c, - main/minimime/tests/messages/test2.txt, - main/minimime/tests/messages/test3.txt, - main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c, - main/minimime/tests/messages/test4.txt, - main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h, - main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c, - main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt, - main/minimime/mimeparser.l, main/minimime/mm_context.c, - main/minimime/mimeparser.tab.h, main/minimime (added), - main/minimime/mm_warnings.c, main/minimime/mm_queue.h, - main/minimime/tests/messages, include/asterisk/autoconfig.h.in, - main/minimime/mimeparser.y, Makefile.moddir_rules, - main/minimime/sys, main/minimime/tests/Makefile: To be able to - achieve the things that we would like to achieve with the - Asterisk GUI project, we need a fully functional HTTP interface - with access to the Asterisk manager interface. One of the things - that was intended to be a part of this system, but was never - actually implemented, was the ability for the GUI to be able to - upload files to Asterisk. So, this commit adds this in the most - minimally invasive way that we could come up with. A lot of work - on minimime was done by Steve Murphy. He fixed a lot of bugs in - the parser, and updated it to be thread-safe. The ability to - check permissions of active manager sessions was added by Dwayne - Hubbard. Then, hacking this all together and do doing the - modifications necessary to the HTTP interface was done by me. - -2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard@digium.com> - - * UPGRADE.txt: clarified a sentence in the format_wav section - - * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and - plan to remove GAIN code from trunk - -2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: When a station picks up a trunk that was on - hold, make the hints reflect that nobody has the trunk on hold - anymore. - - * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459, - reported by francesco_r, fixed by me) * The original behavior was - that if one station put a call on hold, another one picked it up, - and then hung up, the code would still consider the call on hold - by the first station, so the trunk would not be hung up. However, - to better comply with what most people seem to expect it to - behave, it will now hang up the trunk. * Fix a problem with - "barge=no". This was only intended to prevent people from joining - calls that are in progress. However, it also prevented other - people from picking up a call that was on hold. This has been - fixed. * When there are no active stations on a trunk and it is - on hold, the code now indicates the HOLD and UNHOLD conditions to - the trunk channel. This allows music on hold to be played to the - trunk when it is on hold. - -2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we check the faxdetect option - before doing fax processing - - * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 - lines There should only be one code path for doing DTMF - conditionals on channels. This fixes it. ........ - -2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) - | 2 lines remove undocumented 'cardsmode' parameter and stop - searching for transcoders during reload() ........ - -2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c, apps/app_speech_utils.c, - include/asterisk/speech.h: Add support for returning different - types of results (ie: NBest). - -2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard@digium.com> - - * formats/format_wav.c: modified default GAIN for issue 5823, - thanks jrwalliker - -2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf@digium.com> - - * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added - some clarification to the example configs for CDRs, on how to - select a backend. Also, made cdr-csv the default if you 'make - samples', and no other changes. - -2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 - lines Just because we can't find the voicemail configuration - file, doesn't mean that the module failed to load. The user could - be using realtime. Issue #9473 ........ - -2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell@digium.com> - - * main/http.c: Add the MIME type for gif by request from Pari - -2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 - lines Only unlock our pvt and net locks if we are actually going - to try to lock the owner again. (issue #9472 reported by zoa) - ........ - -2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 60134 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | - 6 lines It is valid to redirect channels via the manager - interface that are not in the UP state. Instead of checking for - that to prevent to ensure a dead channel doesn't get redirected, - just use the ast_check_hangup() API call. (issue #9457, reported - by Callmewind, patch by me) (related to issue #8977) ........ - - * channels/chan_sip.c: Add a Content-Length of 0 to the response - built by transmit_response_with_unsupported(). (issue #9454, - reported by makoto, fixed by me) - - * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | - 4 lines Fix the return value of handle_common_options() so that - it always properly indicates whether it handled the option or - not. (issue #9455, reported by Netview, fixed by me) ........ - - * apps/app_meetme.c: Fix a problem where if a trunk was hung up - while it was on hold, all of the hints would reflect the line - still on hold, even though it should reflect that it is back to - not in use. (issue #9459, reported by francesco_r, fixed by me) - -2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Don't clash when a person both speaks - and uses DTMF. - -2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | - 4 lines Don't attempt to report configuration errors in - build_user(). oej pointed out that for a "friend" entry, this - won't work, because all user options are valid for peers, but not - the other way around. ........ - - * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | - 3 lines Make chan_sip report when it encounters an unknown - option. (issue #9440, reported by nightcrawler) ........ - - * /, main/app.c: Merged revisions 59886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | - 5 lines When doing a built-in blind or attended transfer, restore - the ability to use '#' to terminate the number and immediately do - the transfer instead of having to dial the number and just wait - for the feature digit timeout. (issue #8366, xueliangliang) - ........ - - * Makefile: Ensure that menuselect gets executed in dependency - check mode every time you run make. - -2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h: - Merged revisions 59788,59803 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 - lines Use the new sysfs way of mISDN 1.2 to check if a port is NT - or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, - 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ - -2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: - Merged revisions 59623-59624,59639 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | - 1 line we can now make 30 channels on a PRI (before we forgot - chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 - (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ - r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | - 1 line added option which allows us to accept incoming SETUP - Messages without automatically sending Proceeding or Setup - Acknowledge, this is useful with some broken switches and if you - want to Release incoming calls without previously having - acknowledged them. The new option is - noautorespond_on_setup=yes|no default is no, so we don't break - the existing behaviour ........ - -2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 - lines Increase the maximum size for a string of mailboxes to - 1024. (issue #9270 reported by rtucker) ........ - -2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: continue in for-loop should go to the incrementer, - not the test. As per 9435, thanks to marcelbarbulescu - -2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell@digium.com> - - * main/netsock.c, /: Merged revisions 59608 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | - 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock. - This is needed by the patch that went in for issue 7874. - chan_iax2 needs to be able to create socket that is lisetning on - INADDR_ANY, but also be able to bind sockets to specific - addresses. (Thanks to Stevenson on the asterisk-dev mailing list - for explaining why this flag was needed.) ........ - -2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker@digium.com> - - * configure, main/Makefile, acinclude.m4: Add linux-uclibc host - arch..."thingy". Sorry, I don't know what it's called... - -2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, - include/asterisk/cdr.h: several changes via kpflemings review - - * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, - include/asterisk/cdr.h: These mods fix CDR issues from 8221, - 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated - from transfer situations. - - * configs/extensions.conf.sample: A small clarification to keep - bugs from being filed, and confusion from rising, if - clearglobalvars is set, and globals are set in the AEL file. - (9419) - -2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: When building a response to a subscription, the - "from" must be the full Jabber ID. This fixes some problems where - jabber users are not able to add their Asterisk account to their - user list, since they are unable to get Asterisk to approve their - subscription. (issue #8210, reported by caspy, and verified by - bradtem) - -2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 - lines Keep a global array of variables indicating whether certain - conference rooms are in use. This ensures that two people going - into a new dynamic conference when the 'e' option is set don't go - into the same conference room. (issue #8835 reported by eliel) - ........ - -2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell@digium.com> - - * main/rtp.c, /: Merged revisions 59357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | - 5 lines If an error occurs when reading from an RTP socket, and - the error code does not indicate that we should try again, then - return NULL instead of a "null frame". This will prevent Asterisk - from trying over and over again, and eventually causing the - system to crash. (issue #8285, john) ........ - - * channels/chan_iax2.c: When the IAX2 read callback gets called, - return NULL instead of a "null frame". This will cause Asterisk - to hangup the call instead of keep trying whatever it was doing. - Under normal conditions, this function would *never* be called. - However, the author of this patch says an error will occur that - will cause it to get called every 100 thousand calls or so. When - this does happen, it puts the channel in a loop that eventually - brings down the system. So, hangup up the call is certainly a - better alternative. (issue #8286, john) - - * Makefile: Export the GTK2 library and include information to sub - Makefiles. - -2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007) - | 3 lines Issue 9415 - No point to getting a diagnostic field if - we aren't doing anything with the information. (Plus, it tends to - crash the Postgres ODBC driver.) ........ - -2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Another crash that I thought we had fixed already - - Issue 9396 - - * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007) - | 2 lines Oops ........ - - * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007) - | 2 lines Fix a few remaining bad mmap(2) return values ........ - -2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell@digium.com> - - * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) | - 3 lines Fix the check of the return value from mmap(). Thanks to - Corydon for catching this one. ........ - - * apps/app_directory.c: Fix app_directory to actually compile with - ODBC_STORAGE, and update the code to the latest res_odbc API. - - * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used. - The Makefile did not properly ensure that this information got - copied from what was selected for app_voicemail. (issue #9224) - - * channels/chan_sip.c: Fix the check that ensures that the CHANNEL - function's first argument is "rtpqos". Thanks, Corydon. :) - -2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes - asterisk), kpfleming pointed on asterisk-dev, that DECLINE in - this case the proper thing to do. This change now has it doing - the proper thing. - -2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | - 4 lines Fix the use of the "sourceaddress" option when "bindaddr" - is set to 0.0.0.0 instead of having each interface explicitly - listed. (issue #7874, patch by stevens) ........ - - * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS - function to just be additional parameter of the CHANNEL function. - This way, it will be possible for other RTP based channel drivers - to expose this information in the future. - -2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 - Mär 2007) | 1 line fixed #9355 ........ - -2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Oops, this should be case insensitive - -2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes - asterisk). I turned a duplicate context from a WARNING to an - ERROR. Now you get a module load failure, and asterisk just - exits. That's better than a crash, right\? - -2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Change this to a single dp function to make - oej happy. - -2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf@digium.com> - - * main/config.c: Fix for 9257; by eliminating the globals in - main/config.c, we make it thread-safe, which is a minimum - requirement. - -2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Add ability to specify no timeout. This - means as soon as the prompt is done playing it moves on to the - next priority. - -2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Somehow the code for building the email for - voicemail got out of sync. This change makes a few tweaks to get - 1.4 in sync with trunk. (issue #9301) - - * apps/app_meetme.c: Fix some codec negotiation problems when - CallerID support is not enabled in SLA. (issue #9308, reported by - twilson) - -2007-03-26 18:13 +0000 [r59213] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Make SpeechBackground obey the digit - timeout value. - -2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Rename the new dialplan functions to match - the variable name - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The - AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in - some because they get set in sip_hangup. So, there are common - situations where the variables will not be available in the - dialplan at all. So, this patch provides an alternate method for - getting to this information by introducing AUDIORTPQOS and - VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, - with some testing by blitzrage) - -2007-03-26 17:38 +0000 [r59206] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, - pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE, - and STANDALONE_AEL - -2007-03-26 15:25 +0000 [r59202] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure, - include/asterisk/autoconfig.h.in, channels/misdn/Makefile, - channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2 - provides a dsp pipeline for i.e. echo cancellation modules, make - chan_misdn use it. * add a check for linux/mISDNdsp.h to - configure.ac and update the autogenerated files: 'configure', - 'autoconfig.h.in' (the 'configure' script was not in sync with - the latest configure.ac, so the diff is a bit bigger than - expected). - -2007-03-26 15:16 +0000 [r59200] Joshua Colp <jcolp@digium.com> - - * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the - aelparse binary! DONT_OPTIMIZE should now work once again. - -2007-03-24 01:39 +0000 [r59195] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 - lines Only try to handle a response if it has a response code. - (ASA-2007-011) ........ - -2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy <murf@digium.com> - - * /: blocking out the fix in 59187... already incorporated here - - * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 - line Added a few words in the Macro doc strings about the - behavior of macros with hangups (et al.), as per 9337 ........ - -2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't allow string input to overrun the - buffer to hold it (ASA-2007-010) - - * channels/chan_misdn.c: remove variables that are no longer used - (--enable-dev-mode is good, developers should be using it) - -2007-03-22 14:40 +0000 [r59145] Steve Murphy <murf@digium.com> - - * utils/Makefile: The stuff in utils was compiling with -O6 even if - DONT_OPTIMIZE is set in menuconfig. Added the include to fix that - -2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp <jcolp@digium.com> - - * main/http.c: Add svg mimetype for pari. - - * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 - lines Indicate the filename changed when it is changed. (issue - #9311 reported by jsmith) ........ - - * channels/chan_sip.c: Until we can do media level parsing for - sendrecv/etc just use the first value found. This crept up when a - phone was offered audio+video and returned an inactive video - stream. chan_sip thought the phone said to put the person on hold - but that was totally wrong. (issue #9319 reported by benbrown) - -2007-03-20 21:04 +0000 [r59078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/logger.c: Fix defines for inline stack backtraces (only used - by developers anyway) - -2007-03-20 20:42 +0000 [r59076] Joshua Colp <jcolp@digium.com> - - * channels/iax2-parser.c: Copy len variable as well, should fix - remaining IAX2 DTMF issues. - -2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy <murf@digium.com> - - * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should - return it to its previous, untouched, state. - - * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h: - The fix for the AEL <<security hole>> (bug 9316) is here... - -2007-03-20 13:16 +0000 [r59064] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h: Merged revisions - 58849-58850,59062-59063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | - 1 line added method standard_dec for dialing out on groups, to - avoid conflicts, which caused issues with some ISDN providers - ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 - Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | - crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line - avoid sending a disconnect when we already received one. ........ - r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | - 1 line modified a loglevel ........ - -2007-03-19 Jason Parker <jparker@digium.com> - - * Asterisk 1.4.2 released. - -2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Oops, this should have been a %d all along - -2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp@digium.com> - - * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295 - reported by ajohnson) - -2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported - via -dev list) - -2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return - code 0 (reported by qwerty1979) - -2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke@btwtech.com> - - * apps/app_followme.c: Don't return a non-zero return code if the - profile doesn't exist, to match what the documentation says it - already does. (#9307 Reported by kkiely) - -2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp@digium.com> - - * apps/app_page.c: Wait for the async thread to exit when hanging - up all of the paged phones under all circumstances. (issue #9181 - reported by PhilSmith) - -2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: fix a couple SLA documentation - references - - * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex - (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added), - doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added), - doc/channelvariables.txt (added), doc/ael.txt (added), - doc/billing.tex (removed), build_tools/prep_tarball, - doc/callingpres.txt (added), doc/enum.txt (added), - doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added), - doc/cdrdriver.tex (removed), build_tools/make_buildopts_h, - doc/security.txt (added), doc/imapstorage.txt (added), - doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed), - doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac, - doc/iax.txt (added), doc/ael.tex (removed), - doc/channelvariables.tex (removed), doc/enum.tex (removed), - doc/security.tex (removed), doc/math.txt (added), Makefile, - doc/imapstorage.tex (removed), doc/privacy.tex (removed), - doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt - (added), apps/app_voicemail.c, doc/cliprompt.txt (added), - doc/chaniax.txt (added), doc/app-sms.txt (added), - doc/ast_appdocs.tex (removed), doc/realtime.tex (removed), - doc/ices.txt (added), doc/dundi.tex (removed), - doc/linkedlists.txt (added), doc/queuelog.txt (added), - doc/extconfig.txt (added), doc/radius.txt (added), - doc/cliprompt.tex (removed), doc/chaniax.tex (removed), - doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex - (removed), doc/ices.tex (removed), doc/asterisk.tex (removed), - doc/queuelog.tex (removed), doc/configuration.txt (added), - doc/asterisk-conf.txt (added), doc/sla.pdf (added), - doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt - (added), doc/mp3.tex (removed), doc/configuration.tex (removed), - doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added), - doc/channels.txt (added), doc/ip-tos.tex (removed), - doc/extensions.txt (added), doc/queues-with-callback-members.txt - (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added), - doc/misdn.txt (added), doc/manager.txt (added), - doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), - doc/billing.txt (added), doc/localchannel.txt (added), - doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt - (added), doc/00README.1st (added): Making these documentation - changes in the 1.4 branch upset various people, so these chanes - will only be done in the trunk. - - * build_tools/prep_tarball: Add the --pdf option to the usage of - rubber in prep_tarball - - * Makefile, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add - configure script checking for GTK2 and some additional Makefile - targets to support gmenuselect - -2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match - common syntax and update the resulting appdocs TeX file - -2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell@digium.com> - - * doc/asterisk.tex: add a link to the rubber homepage - -2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_setcdruserfield.c, main/pbx.c, - apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c: - Expand deprecation warnings from simply warning on use to the - builtin documentation. - -2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell@digium.com> - - * doc/asterisk.tex, Makefile: Add Asterisk version information to - the generated PDF - - * build_tools/prep_tarball: have prep_tarball attempt to build - asterisk.pdf - -2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_realtime.c: Function works fine, but the documentation - is backwards. - -2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell@digium.com> - - * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex - (added), doc/freetds.txt (removed), doc/odbcstorage.txt - (removed), configure, doc/sla.tex, doc/cygwin.txt (removed), - doc/model.txt (removed), doc/channelvariables.txt (removed), - doc/ael.txt (removed), doc/billing.tex (added), - doc/callingpres.txt (removed), doc/enum.txt (removed), - doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed), - doc/cdrdriver.tex (added), build_tools/make_buildopts_h, - doc/security.txt (removed), doc/imapstorage.txt (removed), - doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added), - doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac, - doc/iax.txt (removed), doc/ael.tex (added), - doc/channelvariables.tex (added), doc/enum.tex (added), - doc/security.tex (added), doc/math.txt (removed), Makefile, - doc/imapstorage.tex (added), doc/privacy.tex (added), - doc/realtime.txt (removed), doc/dundi.txt (removed), - doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt - (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), - doc/ast_appdocs.tex (added), doc/realtime.tex (added), - doc/ices.txt (removed), doc/dundi.tex (added), - doc/linkedlists.txt (removed), doc/queuelog.txt (removed), - doc/extconfig.txt (removed), doc/radius.txt (removed), - doc/cliprompt.tex (added), doc/chaniax.tex (added), - doc/hardware.txt (removed), doc/mp3.txt (removed), - doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex - (added), doc/queuelog.tex (added), doc/configuration.txt - (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf - (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added), - doc/h323.txt (removed), doc/mp3.tex (added), - doc/configuration.tex (added), doc/asterisk-conf.tex (added), - doc/jitterbuffer.txt (removed), doc/channels.txt (removed), - doc/ip-tos.tex (added), doc/extensions.txt (removed), - doc/queues-with-callback-members.txt (removed), doc/apps.txt - (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt - (removed), doc/manager.txt (removed), doc/jitterbuffer.tex - (added), doc/extensions.tex (added), doc/billing.txt (removed), - doc/localchannel.txt (removed), - doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt - (removed), doc/00README.1st (removed): Merge changes from - svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc - directory into a single LaTeX formatted document so that we can - generate a PDF, HTML, or other formats from this information. * - Add a CLI command to dump the application documentation into - LaTeX format which will only be include if the configure script - is run with --enable-dev-mode. * The PDF turned out to be close - to 1 MB, so it is not included. However, you can simply run "make - asterisk.pdf" to generate it yourself. We may include it in - release tarballs or have automatically generated ones on the web - site, but that has yet to be decided. - -2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Don't assume that the pvt structure will - still exist after calling schedule_delivery as it may not. (issue - #9278 reported by fmachado) - -2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Some people like to put "limitonpeer" - instead of "limitonpeers" in their configuration. While we're at - it, support "limitonpeerz" and "limitonpeerssssss". (inspired by - issue #9172) - - * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the - examples section - - * doc/security.txt, /: Merged revisions 58896 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | - 3 lines Add a note to the security file that the Asterisk CLI and - log files may contain sensitive information, and that people - should keep this in mind. ........ - - * configs/sla.conf.sample, apps/app_meetme.c: By default, don't - attempt to do any CallerID handling at all with SLA because it is - known to not work properly in some situations. However, add an - option to enable it for those that would like to use it anyway. - The short story behind this is that to properly handle CallerID - with SLA, we need the ability to change the CallerID on an - existing call, and we are not ready to handle that. - -2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Issue 9162 - - pbx_substitute_variables_helper assumes the buffer is initialized - to all zeroes. This fixes a case where it wasn't. - -2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Ensure that the blinky lights show that the - trunk stopped ringing when the trunk hangs up before a station - has answered it. (issue #9234, reported by francesco_r) - - * configs/sla.conf.sample: fix the reference to the SLA - documentation - -2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 - lines Issue #9229 - No port in request URI on register to non - default SIP ports (neelakantan) ........ - - * channels/chan_sip.c: Don't hangup the call on OK or errors on - MESSAGE and INFO inside of a dialog (like video update requests). - - * channels/chan_sip.c: Issue #9251 - Clear From URI from user - attributes (tgrman) - -2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 57034,57523,57753,58558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | - 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com - bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 - 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ - r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | - 1 line fixed another place where the out_cause was hardcoded to - 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 - Mar 2007) | 1 line we can free channel 31 as well, since we can - occupy it ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c: added UU transceiving and - corect handling for rdnis - -2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Allow RFC2833 compensation to compensate for even - stupider implementations by queueing up the end frame at the - start, not the actual end. (issue #8963 reported by AndrewZ) - - * channels/chan_sip.c, configs/sip.conf.sample: Add - matchexterniplocally setting which only substitutes your - externip/externhost setting if it matches the localnet setting. I - know of at least two people who need opposite settings, so I made - it an option! (issue #8821 reported by kokoskarokoska) - -2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a few more places in chan_iax2 where - the ast_frame used for receiving a frame was not properly - initialized. - Interpolating a frame when the jitterbuffer is in - use - decrypting a frame when IAX2 encryption is on - frames in - an IAX2 trunk - - * apps/app_meetme.c: Make the compiler happy and initialize a - variable. - - * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added): - Merge some updates to the SLA documentation. I plan to keep - working on this to explain all of the expected behavior with call - handling, configuration details for specific phones, and other - things. However, I got tired of doing it in plain text, so I - switched to using LaTeX. I have included the PDF version. I - haven't been able to get a nice looking plain text version out of - it yet, but I'm not terribly concerned since this is supposed to - be more of the manual, while the plain text sample configuration - file is the reference. - -2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Fix spelling of unavailable in voicemail - documentation. (issue #9248 reported by tensai) - - * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 - lines If we are unable to lookup the host in a c line we have to - abort, otherwise the previous data is gone and we will - (potentially) have no data when all is said and done. ........ - -2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Hang up the channel that put the call on hold - in the event processing thread to avoid a race condition. Also, - if the station originated the call that it is putting on hold, - don't hang up the trunk if it was the only station on the call - and it is hanging up due to hold and not a normal hangup. - - * channels/chan_zap.c: Add a missing break statement so that - handling the above event does not incorrectly destroy the - channel. (issue #9242, andrew) - -2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Fix segfault (Issue 9236) - -2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Refactor hold handling a bit so that it does - not require keeping the call up when a call is put on hold. - -2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Make early SDP seeding even smarter! We have to check - codecs in the make_compatible function too. (issue #9221 reported - by marcelbarbulescu) - - * main/dsp.c, /: Merged revisions 58388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 - lines Only print out debug message if the definition that makes - the variables shows up was actually defined. (issue #9233 - reported by serginuez) ........ - -2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming@digium.com> - - * main/http.c: this change was not needed; fclose() handles closing - the file descriptor already - - * apps/app_meetme.c: fix a compiler warning, and overwriting 'res' - value - - * main/http.c: fix two cases where HTTP session file descriptors - would not be closed - -2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, configure, configure.ac: If we receive - ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, - tzafrir) Also, update the configure script to make sure that we - don't try to build chan_zap if the installed version of zaptel - does not include ZT_EVENT_REMOVED. - - * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey - Moore) Merged revisions 58242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | - 7 lines Fix a problem where the Asterisk channel name could be - that of the wrong IAX2 user for a call. This is because the first - step of choosing this name is to look for an IAX2 peer that - happens to have the same IP/port number that this call is coming - from and assuming that is it. However, this is not always - correct. So, I have made it change this name after authentication - happens since at that point, we have an exact match. ........ - -2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have) - at least one matching codec before attempting early bridge SDP - seeding. (issue #9221 reported by marcelbarbulescu) - -2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 58164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | - 4 lines If the channels acquired using the manager Redirect - action are not up, then don't attempt to do anything with them. - It could lead to weird behavior, including crashes. (issue #8977) - ........ - -2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 - line Fix for 9220: Eyebeam cannot renew subscriptions for - presence info. Reason: re-SUBSCRIBE requests don't include Accept - headers, which the rfc says are optional (to put it tersely), (it - uses MAY), and luckily, the sip_pvt struct has the format info - stored, so we simply leave it if the format is set, and the - accept header null. ........ - -2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell@digium.com> - - * configs/voicemail.conf.sample: Clarify the documentation of the - dialout and sendvoicemail options. (issue #9000, caio1982 and - serge-v) - -2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 - lines Change error message to proper message ........ - -2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell@digium.com> - - * channels/chan_skinny.c: Return an error of transmit_response is - called without a session. (issue #9002) - -2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Since chan_iax2 does not support reception - of DTMF with duration ensure that it is set to 0 on the frame. - (issue #8521 reported by gdhgdh) - - * apps/app_meetme.c: Don't create a listen channel and record the - conference unless the option is turned on. (issue #9204 reported - by francesco_r) - - * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 - lines Make create_dirpath use our standard for return values. -1 - is failure, 0 is success. (issue #9205 reported by ballares) - ........ - -2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57825 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 - line Fixed a typo introduced via 9156 (either the gotos or their - doc strings are wrong) ........ - -2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Don't allow a NULL pointer to reach - ast_frdup. (issue #9155 reported by cmaj) - - * res/res_jabber.c: Don't reference a potentially NULL pointer. - (issue #9199 reported by klolik) - - * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198 - reported by edgreenberg) - -2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2, - pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7: - Updated the regression tests - -2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) - | 2 lines Memory leak of a list, if call recording was abandoned - ........ - -2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/say.c: submitted patch for Georgian language, issue 9010, - submitted by Alexander Shaduri - -2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: add missing configuration template. - Thanks to Lacy Moore on asterisk-users for pointing this out\! - -2007-03-02 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.1 released. - -2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Update the check that is used to - determine whether zaptel transcoder support is present. The - interface has changed. - -2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 - lines If a SIP message comes in and goes to a method handler that - requires additional values that may not be present then send back - an error. ........ - -2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57458 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 - line further refinement in wording of goto documentation, as per - 9156, goto not proceeding to next instruction ........ - - * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes - right, but 9184 points out the problem-- the escape is removed by - pbx_config, and pbx_ael should also, before sending it down into - the pbx engine. Also, you have to insert it back in, if you are - generating extensions.conf code from the AEL. - -2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com> - - * main/file.c: Return the correct digit that interrupted the - stream. This fixes exiting the Background application when using - the m option. (issue #9176, mjagdis) - - * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt, - include/asterisk/channel.h: Merge changes from - svn/asterisk/team/russell/sla_updates * Originally, I put in the - documentation that only Zap interfaces would be supported on the - trunk side. However, after a discussion with Qwell, we came up - with a way to make IP trunks work as well, using some things - already in Asterisk. So, here it is, this now officially supports - IP trunks. * Update the SLA documentation to reflect how to setup - IP trunks. * Add a section in sla.txt that describes how to set - up an SLA system with voicemail. * Simplify the way DTMF - passthrough is handled in MeetMe. * Fix a bug that exposed itself - when using a Local channel on the trunk side in SLA. The - station's channel needs to be passed to the dial API when dialing - the trunk. * Change a WARNING message to DEBUG in channel.h. This - message is of no use to users. - -2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 57317 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar - 2007) | 2 lines Don't even attempt to optimize things when a - proxy channel is involved. It will just explode in weird and - unexplaineable ways. (issue #9175 reported by - clegall_proformatique) ........ - -2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. - -2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla - docs - - * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes - from svn/asterisk/team/russell/sla_updates * Add support for - private hold. By setting "hold=private" for a trunk, only the - station that put the call on hold will be able to retrieve it - from hold. Also, by setting "hold=private" for a station, any - call that station puts on hold can only be retrieved by that - station. - - * apps/app_meetme.c: Minor formatting change - - * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from - svn/asterisk/team/russell/sla_updates * Add support for the - "barge=no" option for trunks. If this option is set, then - stations will not be able to join in on a call that is on - progress on this trunk. - -2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57118 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 - line a small documentation update, to reflect reality in the goto - doc strings, as per 9156, Goto does not proceed to next prio if - jump fails ........ - -2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb - 2007) | 2 lines Fix a few more issues with the agent logoff CLI - command. (issue #9123 reported by arbrandes) ........ - -2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of - changes from svn/asterisk/team/russell/sla_updates * Add support - for station ring delays. Ring delays can be set globally for a - station or for specific trunks on the station. * Fix a few bugs - in existing code. * Restructure and Reorganize code to improve - readability and maintainability. * Improve formatting of the "sla - show (trunks|stations)" CLI commands. - -2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Picky compiler... - - * apps/app_speech_utils.c: Better handle timeouts when the - individual speaks after everything has been played but before the - timeout ends. - -2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: I was surprised that I had not yet downgraded - missing goto targets and macro call defs to a warning, in case - they are in extensions.conf; I rectified this problem. Also, A - goto in a macro to a target in a catch block was not being found; - I fixed this too; the cause was that I needed to treat catch - statements like an extension in the find_match code. - -2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Fix voicemail email attachments. I missed - the conversion of one of the line endings and there was an extra - one where it should not have been. (issue #9128) - -2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky, - picky... show deprecation warning in application help, too - (reported via list) - -2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com> - - * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where - if a device was not specified in alsa.conf, then we just use the - system default, instead of creating our own default of hw:0,0. - (issue #9139) - -2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 - lines Obey the clearglobalvars option in extensions reload (or - dialplan reload depending on your version). (issue #9146 reported - by ramonpeek) ........ - -2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a crash in my last change to - iax2_indicate(). (issue #9150) - -2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com> - - * apps/app_record.c: Update app_record documentation to use new CLI - command, core show file formats. (issue #9151 reported by junky) - - * main/pbx.c: Use ast_strlen_zero to see if the language and/or - context argument is not present for Background instead of just - checking if it is NULL. (issue #9141 reported by mjagdis) - -2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Do more complete locking of the - chan_iax2_pvt struct in the indicate callback. (Problem brought - up by Ben Smithurst on the asterisk-dev list) - -2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Allow both of the show version files and core - show file versions CLI commands to work. (issue #9135 reported by - mvanbaak) - -2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Move a comment to be in the correct struct. - -2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/channel.c, /: Merged revisions 56684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) - | 3 lines Issue 9130 - If prev is the last item on the channel - list, then evaluating additional conditions (e.g. name prefix) - will cause a NULL dereference. ........ - -2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure to set a speeddials parent on - creation. Don't crash if hold is pressed when no call is active. - Don't return in places that we shouldn't.. - -2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: update to match zaptel 1.4 API change that - was committed a few minutes ago - -2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com> - - * main/asterisk.c, /: Merged revisions 56504 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | - 8 lines Fix up a couple more signal handlers to not do bad things - that could cause various undesirable results. The other day, I - made Asterisk deadlock by hitting Control-C because of a bad - signal handler. Now, signal handlers just set a flag and write to - an alert pipe for the flag to be handled. Then, there is another - thread that is monitoring for these flags. If being run in - console mode, it is just the main thread. If Asterisk is in the - background, a thread is created to do it. ........ - -2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com> - - * main/sched.c: Change log notice to debug. It is possible for a - scheduled item to execute and be deleted at close to the same - time and unavoidable. If this happens this message creeps up. - -2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | - 4 lines Don't destroy mutexes before unregistering all of the - entry points from the core. Also, fix a potential memory leak - from not destroying the locks for all of the possible call - numbers (about 32k of them). ........ - -2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_version_h: build special version strings for - AADK/S800i builds - -2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: The IMAP storage code uses the same code to - build the email that is used when voicemail is sent via email - using something like sendmail. In the patch from bug 8033 to fix - various IMAP storage problems, the line endings in the email file - were changed in the code from "\n" to "\r\n". However, this - breaks sending regular voicemail to email. So, this change - conditionally sets line endings to "\r\n" only if IMAP_STORAGE is - enabled. (issue #9128, patch by jarjarbinks, modified by me to - not break IMAP storage) - -2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c, - doc/sla.txt: Merge changes from team/russell/sla_updates. This - batch of changes to the SLA code does a few different things. * I - made the SLA code event driven instead of having to act in a lot - of busy loops while dialing things to wait for state changes. - This makes the code more efficient and readable at the same time. - * I have implemented a couple of new features. The first is - inbound trunk ringing timeouts. This is an option that defines - how long to let an incoming call on a trunk to ring. * I have - also implemented ring timeouts for stations. They may be - specified for the entire station, meaning it is how long to let - the station ring before giving up. You can also specify a ring - timeout for a specific trunk on a station. So, you can say that - you only want a specific station to ring 5 seconds if it is line1 - ringing, but otherwise, there is no timeout. - -2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 - lines Only change the original or clone channel if it's the - channel behind the proxy channel, not if it's just a regular - bridged channel. ........ - -2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: Update OSP documentation for v1.4. - -2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Move message from verbose to debug - -2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com> - - * sounds/Makefile: updated the sound tarball versions in Makefile - -2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Restructure a little bit of code to reduce - nesting. There is no functionality change here. - - * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | - 3 lines If we receive a frame that is not in any of the - negotiated formats, then drop it. (potentially issue #8781 and - SPD-12) ........ - -2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com> - - * main/cli.c: Print out deprecation notice on usage output of CLI - commands. (issue #8925 reported by blitzrage) - -2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com> - - * main/loader.c: disable unloading of embedded modules... there is - a fundamental problem with doing so that will not be fixed in - this version of Asterisk due to its invasiveness - -2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 - lines Change naughty warning message to provide useful - information. If a write now fails on a channel in meetme it will - tell you the channel name instead of spitting out the wrong error - message. ........ - -2007-02-21 20:27 +0000 [r55954] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c: Fix locking issue, and accept - "transport-accept" as a valid accept message. This should solve - issues 8970 and 8503. - -2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Simplify the last change to app_meetme, and - move the call to dispose_conf() up into the block where we know a - conf exists. - -2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Only dispose of the conference if one was - created. - - * apps/app_speech_utils.c: Only start playing the next file if we - have not been quieted. - - * channels/chan_sip.c: Add a flag that indicates whether a SIP - dialog is an outgoing call or not. SIP_OUTGOING originally did it - but it was repurposed to the direction of the last transaction, - which can cause update_call_counter to falsely decrease the wrong - counters. (please don't hurt me oej) (issue #8943 reported by - mdu113) - -2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/make_version: Merged revisions 55868 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 - Feb 2007) | 2 lines use new tag version script ........ - -2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly - after transfer (decrement inuse early on transferer's call leg) - -2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found. - Issue 7764, patch by sailer - -2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Improve the reference counting to fix bugs - where people report seeing conferences listed that have no - members. (issue #9073) - -2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Better handle dropped IMAP connections. - (issue #9054 reported by bsmithurst) - - * channels/chan_sip.c: Return behavior I removed. I did not - remember that you could just add a localnet entry to make it - work. - - * channels/chan_sip.c: Don't test our own address against the - localnet settings. At least one person has had issues as a result - of this from #7051 so I'm reversing it. (issue #8821 reported by - kokoskarokoska) - - * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb - 2007) | 2 lines Defer clearing callback information if channels - are up until they are hung up. This ensures the hangup process - goes smoothly and no channels get hung in limbo. (issue #8088 - reported by kebl0155) ........ - -2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com> - - * main/http.c: Add the Asterisk version information to the Server - header in HTTP responses. (requested by Pari) - - * include/asterisk/manager.h: Increase the maximum number of - manager headers to 128, at the request of Pari. - -2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free - with strdupa (thanks file) 55555! - -2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: Change the formatting of sla.conf.sample - to make it more readable. (issue #9112, blitzrage) - -2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net> - - * res/res_jabber.c: - Not sending arguments to an application is - not "out of memory" - Making error messages a bit more clear - -2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) - | 2 lines forcename and forcegreetings options should check to - see if the recording already exists ........ - -2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com> - - * channels/chan_iax2.c: Changed iax2 process thread to detached to - correct memory leak due to left over thread context on thread - exit. Modified module unload process to avoid deadlocks on - pthread cancels - -2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net> - - * /, apps/app_record.c: Merged revisions 55277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 - lines Documentation update (#9053, jsmith) ........ - - * /: Block patch that was made only for 1.2 (already implemented in - 1.4 and trunk) - -2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Add missing membername option to AddQueueMember - documentation. (issue #9088 reported by seanbright) - -2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an issue where callerid would not be - displayed on some phones. Issue 8995, initial patch and research - done by wedhorn - -2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 - lines Answer the channel before recording privacy information. - (issue #8926 reported by lmamane) ........ - - * apps/app_queue.c: Make the 'i' option of Queue actually work. - (issue #8986 reported by utis) - - * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 - lines Allow chan_sip to handle attended transfers from a SIP - phone that is sitting behind chan_agent. Yes folks, all it took - was one line of code. (issue #8784 reported by pzieba) ........ - -2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: If the - pg_config application is found, but there is probably executing - it, then consider postgres unavailable. (issue #8637) - - * codecs/gsm/Makefile: Filter out yet another architecture that - does not work with the optimizations in the built-in libgsm. - (issue 8637, ovi) - - * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged - revisions 55005 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | - 9 lines Revert the change I did in revisions 54955, 54969, and - 54970, in 1.2, 1.4, and trunk. I decided that once a conference - is created from meetme.conf, it is acceptable behavior that the - pin can not be changed until the conference goes away. I also - added a note in meetme.conf to describe this behavior. We still - have another issue in 1.4 and trunk where some conferences with - no users don't go away. That is the real bug that needs to be - addressed here. ........ - -2007-02-16 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb - 2007) | 2 lines Do not send indications through ast_indicate in - chan_agent but instead go directly to the technology. This way - when indications are emulated they happen on the Agent channel - and do not screw up formats on the channels. (issue #8439 - reported by punkgode) ........ - -2007-02-16 21:12 +0000 [r54969] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | - 5 lines For conferences that are configured in meetme.conf, check - the configuration file every time someone joins the conference - instead of only when the conference is first created. This is to - ensure that changes to the pin numbers in the config file are - always honored. (issue #9073) ........ - -2007-02-16 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Need to check macro extension as well as macro - context for directed pickup. - -2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by - default. It was set to enabled in pbx.c. However, if the option - was not present in extensions.conf, then pbx_config.c would set - it back to disabled. - - * res/res_features.c: Clean up a few coding guidelines issues - - spaces to tabs, use sizeof() to pass the size of a static buffer, - add spaces ... - -2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com> - - * main/asterisk.c: Clarify a restart message. It's silly, but the - reporter had a very valid point. Issue 9079 - -2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Allow directed pickup to pick up the real - context instead of the macro context if a Macro is used. (issue - #8984 reported by jamesb63) - -2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7541 - Handle multipart attachments - to SIP messages - even if boundary is quoted. - - * /, res/res_agi.c: Merged revisions 54771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 - lines Issue #9069 - If we open with TH we should not close with - /TD. (seanbright) ........ - -2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Don't let dtmf leak over into the engine - and let it skew the results... also give DTMF results priority. - (issue #9014 reported by surftek) - - * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 - lines Use a separate variable to indicate execution should - continue instead of the return value. (issue #8842 reported by - pluto70) ........ - - * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue - #9068 reported by mhardeman) - -2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net> - - * /: Block patch only needed in 1.2 - -2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 - lines When handling glare on a PRI, move the requested channel - rather than hang up the old one. Fix for 8957 and 9011. ........ - -2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees - with it's placement, feel free to change it. (issue #9045 - reported by gork) - -2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove a couple of leftover debug messages - - * include/asterisk/devicestate.h: Fix the documentation on the - return values from device state provider registration and - deletion. - - * channels/chan_sip.c: If we fail to create the SIP socket, then - return -1 from reload_config() so that load_module() will return - AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get - spammed with error messages every time chan_sip tries to send a - message. - -2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net> - - * /: Blocking patch for 1.2 only - -2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com> - - * main/dial.c, include/asterisk/dial.h: Change - ast_set_state_callback() to ast_dial_set_state_callback() - - * main/dial.c, apps/app_meetme.c, apps/app_page.c, - include/asterisk/dial.h: - Add the ability to register a callback - to monitor state changes in an asynchronous dial operation. - - Rename the various references to "status" to "state" in the dial - API - -2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Make the --without-oss argument work. - (issue #9026 reported by puzzled) - -2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com> - - * configs/users.conf.sample: Fix a typo where "vmpassword" should - be "vmsecret" - -2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Fix VLDTMF reception - - * apps/app_echo.c: Much simpler than previous one ;-) - - * main/channel.c: Provide correct DTMF duration - - * main/cli.c: Bring deprecated 'debug channel <x|all>' command back - -2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac, acinclude.m4: don't display the - --with-imap message unless --with-imap was specified without a - path use '-n' instead of '! -z' for tests - -2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Add some output for "show application - SLAStation/SLATrunk" - - * channels/chan_sip.c: Change some text to properly state "On - Hold", which was already done in trunk. - - * configs/sla.conf.sample, include/asterisk/app.h, - include/asterisk/utils.h, main/dial.c, apps/app_meetme.c, - channels/chan_sip.c, doc/sla.txt (added), - include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge - team/russell/sla_rewrite This is a completely new implementation - of the SLA functionality introduced in Asterisk 1.4. It is now - functional and ready for testing. However, I will be adding some - additional features over the next week, as well. For information - on how to set this up, see configs/sla.conf.sample and - doc/sla.txt. In addition to the changes in app_meetme.c for the - SLA implementation itself, this merge brings in various other - changes: chan_sip: - Add the ability to indicate HOLD state in - NOTIFY messages. - Queue HOLD and UNHOLD control frames even if - the channel is not bridged to another channel. linkedlists.h: - - Add support for rwlock based linked lists. dial.c: - Add the - ability to run ast_dial_start() without a reference channel to - inherit information from. - - * apps/app_echo.c: When the Echo() application receives the digit - '#', echo that back as well. Since we already sent the BEGIN - frame for that digit, it makes sense to send the END as well. - -2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_gtalk.c: another dependency - - * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c, - funcs/func_odbc.c, res/res_adsi.c: add some inter-module - dependencies - - * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk - scripts to work when both MODULEINFO and MAKEOPTS are present in - a source file - -2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Temporarily change musicclass on channel to one - specified in Dial so that the 'm' option functions properly. - (issue #8969 reported by christianbee) - -2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com> - - * doc/imapstorage.txt, configure, configure.ac: clarify the fact - that voicemail IMAP storage cannot be built against a distro's - binary c-client library package (at least not at this time) - -2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net> - - * main/acl.c: Don't output debug unless we asked for it - -2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Fix timeout issue when utterance is - longer then timeout itself. - -2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/loader.c: Issue 9007 - Mutex not released on early return - - * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) - | 2 lines Issue 9003 - If fullname is empty, quote() passes back - "\"" ........ - -2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com> - - * main/db1-ast/Makefile: When building libdb1.a, put the additional - flags needed at the beginning of ASTCFLAGS, instead of at the - end. This way, we ensure that we find the local headers first - before accidentally trying to use headers that exist in locations - specified in the ASTCFLAGS passed from the main Makefile. (issue - #8637, ovi) - - * main/Makefile: The clean target actually needs to run "distclean" - on editline. This is because we need to make sure that its - configure script gets executed again, because the CFLAGS we want - to pass to editline may have changed. - -2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: We can not reliably do P2P bridging with DTMF passing - back with compensation if we need to listen for DTMF frames. - (issue #8962 reported by caio1982) - -2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com> - - * main/rtp.c: When parsing the NTP timestamp in a sender report - message, you are supposed to take the low 16 bits of the integer - part, and the high 16 bits of the fractional part. However, the - code here was erroneously taking the low 16 bits of the - fractional part. It then shifted the result 16 bits down, so the - result was always zero. This fix makes it grab the appropriate - high 16 bits, instead. (issue #8991, pointed out by - andre_abrantes) - -2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c: Directly load say.conf in load_module - instead of calling the reload function. (issue #8946 reported by - junky) - - * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 - lines Fix a few potential memory leaks with realtime users and - peers. (issue #8999 reported by bsmithurst) ........ - -2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) - | 2 lines Issue 7440 - Macro called from Macro from the h - extension exits prematurely ........ - -2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 52843 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | - 1 line fixed some possible segfaults. also fixed an very - important bug which occurs on high load (when calls are very fast - generated) ........ - -2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_jabber.c: Text fix for jabber reload command (reported by - bkruse via IRC) - - * main/manager.c, /: Merged revisions 53245 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) - | 2 lines Issue 8987 - Status could return two responses - (mnicholson) ........ - -2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Formatting - -2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c: Ensure say_cfg is NULL when the module is - loaded. (issue #8946 reported by junky) - - * apps/app_playback.c: Unregister Playback CLI commands as well as - dialplan application. (issue #8946 reported by junky) - -2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add some comments on queue system behaviour - and how it affects the SIP channel - -2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make SIPDtmfMode application work with - recent capability changes, and also fix an RTP stack issue when - the auto option was used. (issue #8972 reported by mdu113) - -2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | - 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when - the dial application exits early because of invalid arguments - instead of just leaving it empty. (issue #8975) ........ - -2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string - because due to compatibilities with CS1000 reported at - www.voip-info.org - -2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com> - - * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a - warning to the console that things might possibly be - misconfigured when queue member's states are still 'Not in Use' - when we're about to bridge them with a caller from queue. Also, - put some documentation quoted from oej's queues.txt efforts - started in /trunk today. This commit puts #7433 into feedback - state for 1.4, and pending no further negative feedback, it will - finally be closed. - -2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Correct a copy/pasted error message line for RTCP. - - * main/config.c, /: Merged revisions 53117 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 - lines Pass the glob expanded filename to process_text_line so - that error messages contain the actual filename, not the original - include one. (issue #8959 reported by tzafrir) ........ - - * Makefile: Add systemname to asterisk.conf generation per recent - discussions about it. (issue #8968 reported by blitzrage) - -2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct - p2p RTP call setup in SIP. You can enable it in sip.conf, but it - is now considered experimental until we solve the - AST_CONTROL_ANSWER with payload and videocaps stuff. - -2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 - lines Copy noncodeccapability over to the joint variable so that - telephone-event will get transmitted in the sent INVITE. ........ - - * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile - here as well, but it apparently required both dev mode and no - optimizations to creep up. - - * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 - lines Don't negotiate RFC2833 when not configured to do so. - (issue #8799 reported by mdu113) ........ - -2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com> - - * funcs/func_strings.c: Fix the FIELDQTY function to not crash. - (reported by blitzrage and Corydon on IRC) - -2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net> - - * /: Going backwards, blame file. - -2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb - 2007) | 2 lines Return previous behavior of having MOH pick up - where it was left off. (issue #8672 reported by sinistermidget) - ........ - - * funcs/func_strings.c: Make func_strings build under dev mode. - Didn't I do this today already in the berkeley DB? - -2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement - call counter - If it's still set at time of dialog destruction, - make sure we decrement the device call counter properly before we - destroy the dialog - - * apps/app_queue.c: Change debug level for state change message - that is not really informative when debugging app_queue - - * channels/chan_sip.c: Cleaning up the devicestate callback - function - -2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Oops. - - * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) - | 2 lines Bug 8965 ........ - -2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Add missing 'F' letter to getopt so it magically - becomes a valid option. (issue #8960 reported by tzafrir) - -2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) - | 2 lines No wonder FIELDQTY doesn't work with functions... the - documentation in pbx.c was wrong ........ - -2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix silly logic. We really want to write - UDPTL frames out when the call is up. - -2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Add explanation of port= in combination - with defaultip= (thanks jsmith) - -2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: we update the name on any first reply of - our setup - -2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: chan_h323 is very stable, so let it built - by default - -2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: When going on hold have the side that was put on hold - reinvite back to Asterisk. When going off hold have the side that - was taken off hold reinvited back to the other party. - - * main/rtp.c: Add more frame types to forward in the RTP bridge - loops. - -2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com> - - * main/cdr.c, main/manager.c, pbx/pbx_spool.c, - channels/chan_skinny.c, channels/chan_h323.c, main/http.c, - pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c, - main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, - channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: - Merged revisions 53045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | - 3 lines Fix a bunch of places where pthread_attr_init() was - called, but pthread_attr_destroy() was not. ........ - - * apps/app_userevent.c: Remove an extra \r\n from manager user - events. (issue #8955, mnicholson) - - * main/rtp.c, /: Merged revisions 53039 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | - 3 lines Use the proper format string to print unsigned values in - the rtp debug output. (issue #8954, wmis) ........ - - * apps/app_queue.c: Only changed the paused status in an existing - queue member if the paused column exists. - - * apps/app_queue.c: Instead of always creating a realtime queue - member as unpaused, read the "paused" column and use that value - for the paused status of the member. (issue #8949, jmls) - - * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10. - (issue #8363, johnlange) - - * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue - #8942, lters) - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - codecs/codec_gsm.c: When we are checking for a system installed - version of libgsm, we need to check for gsm.h as well. - Furthermore, when checking for this header, it may be located in - a gsm/ sub directory, so check for that, as well. (issue #8773) - - * channels/chan_sip.c: Only set the DTMF flag on the rtp structure - if the DTMF mode is actually RFC2833, not just that it is not - INFO. This makes it get set for inband DTMF as well, which is not - valid. (issue #8936) - - * main/asterisk.c, /: Merged revisions 52903 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | - 9 lines The SIGHUP handler was implemented to allow admins to - send SIGHUP to a running Asterisk process to reload the - configuration. However, doing the actual reload in the signal - handler itself is a very bad thing to do, because the reload - process includes calling non-reentrant functions such as - malloc/calloc/etc. If Asterisk is running in the background, then - the reload will happen immediately. However, if running in - console mode, the reload doesn't work until something is typed at - the console. That sort of defeats the purpose, but I don't see an - easy way to get around it at this point. ........ - -2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Drop the deprecated show commands since the - original ones were changed back. (issue #8937 reported by - PCadach) - -2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Revert reprecation of h.323 gk cycle - command from pre-1.4 version instead of duplicated h323 cycle gk - - * res/res_odbc.c: Don't play with free()'d pointers - - * configure, acinclude.m4: Handle non-standard OpenH323/PWLib - library names - -2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | - 5 lines Fix the extraction of the timestamp from video frames. It - was using the mapping for a mini-frame instead of a video-frame, - which caused it to get invalid data. (issue #8795, mihai) - ........ - -2007-01-29 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan - 2007) | 2 lines Now that filename is part of the structure and - since it comes before postprocess... we have to add it to our - postprocess line. (reported on asterisk-dev by Boris Bakchiev) - ........ - -2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com> - - * main/Makefile: Add a missing quotation mark. This was pointed out - by jcmoore on #asterisk-dev. - - * main/manager.c: Remove a recursive lock of the manager session. - This was pointed out by zandbelt in issue #8711. - -2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_config.c: Argument number correction - -2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com> - - * main/Makefile: ASTLDFLAGS needs to be passed to the editline - configure script as LDFLAGS. (issue #8928, zandbelt) - - * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF - mode translation. P2P bridging can only be used when the DTMF - modes don't match if the core is monitoring DTMF in both - directions. Then, the core will handle the translation. - Otherwise, this bridging method can not be used. (issue #8936) - - * main/manager.c: The session lock can not be held while calling - action callbacks. If so, then when the WaitEvent callback gets - called, then no event can happen because the session can't be - locked by another thread. Also, the session needs to be locked in - the HTTP callback when it reads out the output string. This fixes - the deadlock reported in both 8711 and 8934. Regarding issue - 8711, there still may be an issue. If there is a second action - requested before the processing of the first action is finished, - there could still be some corruption of the output string buffer - used to build the result. (issue #8711, #8934) - -2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Use ast_calloc instead of malloc. - -2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com> - - * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR - backport to 1.4). It was committed to trunk via 7663. But it - wasn't so much an enhancement as a fix for the bad language - output for portuguese in Brazil, so, after a lot of prodding from - patient Brazilians, here is the same fix for 1.4 - -2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Set quota information to 0 when creating a - vm_state. (issue #8924 reported by neutrino88) - -2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com> - - * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in - the last commit to the adaptive jitterbuffer code. - Specifically - indicate to the compiler that the "dropem" variable only needs - one but. - Change formatting to conform to coding guidelines. - -2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com> - - * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with - jitterbuf, whereas it would not complain about, and would allow - itself to be overfilled (per the max_jitterbuf parameter). Now it - rejects any data over and above that size, and complains about - it. - -2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * configure, configure.ac: Suggested change to fix normal usage of - --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing - list) - -2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 - lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log - follow documentation. (issue #7677 reported by amilcar) ........ - - * main/manager.c: Have the manager interface send back an "Already - logged in" message instead of "Invalid/Unknown Command" when the - client authenticates for a second time. (issue #8509 reported by - pari) - - * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 - lines Make the last context entry read in the dominant one. - (issue #8918 reported by pj) ........ - - * main/file.c: Fix core show file formats CLI command. - -2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com> - - * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 - lines Allow dequeueing of frames with negative timestamp by - moving jitterbuffer frames check to jb_next. (issue #8546 - reported by harmen) ........ - - * channels/chan_sip.c: Drop out variables I accidentally put in. - - * channels/chan_sip.c: Decrement onHold count if we are hung up on - and still on hold. (issue #8909 reported by alexh42) - - * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan - 2007) | 2 lines Add another note about audio files being played - back to each bridged party. (issue #8718 reported by ppyy) - ........ - -2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, configs/users.conf.sample: By suggestion - from kpfleming last week, change "vmpassword" to "vmsecret". - - * configure, configure.ac: Remove libnsl as a required lib for - libiksemel to work. This change was already made in the trunk. - (issue #8762) - - * include/asterisk/dial.h: Fix the formatting of doxygen comments - to properly indicate that the comment documents the previous - entity, as opposed to the next one. - -2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com> - - * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 - line updated check_expr via 8322 (refactoring of expression - checking impl); elfring contributed a nice code reorg, I - contributed some time to get it working again, better messages - ........ - -2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com> - - * main/dial.c (added), apps/app_page.c, main/Makefile, - include/asterisk/dial.h (added): Merge in dialing API and the - app_page that uses it. (issue #BE-118) - - * channels/chan_sip.c: Fix changing channel formats when joint - capability changes and there are no audio formats... I didn't - break it originally! (issue #8535 reported by ivoc) - -2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com> - - * configure: rebuild configure script to reflect last chan_h323 - related changes. - -2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: added fix from #8899 - - * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 - Jan 2007) | 1 line fixed the busy problem (dialstatus was not - busy when we called a busy extension) ........ - -2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Show capabilities *and* preference in - general settings in "sip show settings" (reported by Clona/Telio - - Thanks!) - -2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz> - - * acinclude.m4: Allow x64 builds of H.323 (please, rebuild - configure) - -2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 51843 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | - 6 lines Fix an issue related to synchronization of recordings - when using Monitor(). The bug is a miscalculation of the amount - to seek the stream for writing to disk when the number of samples - coming in and out of a channel do not match up. (issue #8298, - #8887, report and patch by guillecabeza, patch files created and - testing done by whoiswes) ........ - - * apps/app_while.c, /: Merged revisions 51828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | - 4 lines Don't set a new value for the END_ variable on the - channel before using the old value. If you do, it will lead to - accessing a memory address that has been free()'d. (issue #8895, - arkadia) ........ - -2007-01-23 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_alsa.c, - channels/chan_gtalk.c, channels/chan_iax2.c: Update channel - drivers to use module referencing so that unloading them while in - use will not result in crashes. (issue #8897 reported by junky) - -2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com> - - * main/manager.c: Fix some bugs in process_message(). The manager - session lock needs to be held when sending some sort of response, - or calling one of the manager action callbacks. This resolves an - issue where people using the GUI would get random crashes when - they start clicking around a lot. (issue #8711, reported and - debugged by zandbelt) - - * main/http.c: Fix setting the default port of 8088 on 64-bit or - big-endian machines. - - * main/manager.c: When traversing the list of manager actions, the - iterator needs to be initialized to the list head *after* locking - the list. Also, lock the actions list in one place it is being - accessed where it was not being done. - -2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com> - - * res/res_features.c: this mod from 8593 (dstchannel in cdr is - empty when transfer call). - - * main/callerid.c: via 8748 (callerid.c loses name when returning - PRIVATE_NUMBER flag), the user suggested this mod, saying it - would allow 'WITHHELD' to appear in the name field, which would - be useful - -2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, - channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | - 6 lines * more additions to make the RESTART message work * added - fix for misdn_call to allow SETUPs with empty extensions, - replaced the strtok_r functions with strsep for that (inspired by - Sandro Cappellazzo, thanks) ........ r50506 | crichter | - 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get - L2 UP, the L1 is UP definitely too, so we set the L1 state up as - well. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c: manually merged r49922 and r50335, because - of conflicts. this commint includes addition of the ISDN RESTART - Message - -2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk - startup if h323 configuration file not found (reported by - mithraen) - -2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only change audio formats on the channel if - we have an audio format to change to. (issue #8535 reported by - ivoc) - - * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan - 2007) | 2 lines Yield before reading from zaptel timing source - under Solaris so that other threads get a chance to do things. - (issue #7875 reported by bob) ........ - -2007-01-22 19:28 +0000 [r51409] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This fixes 8836, according to dnatural - -2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan - 2007) | 2 lines Move filestream creation to Mixmonitor loop. This - will prevent a blank file from being created if no frames ever - pass through to be recorded. (issue #7589 reported by - steve_mcneil) ........ - -2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com> - - * configs/say.conf.sample: Fix Italian numeral support in say.conf - for "_[2-9]00" case. "2131" would've translated to something - along the lines of (pardon my..Italian {or lack thereof}) - "duecentocentotrentuno", which makes no sense at all. - - * configs/say.conf.sample: Fix German language support in say.conf - Properly support 21, 31, 41, 51, 61, 71, 81, and 91. - einundzwanzig has the same format as zweiundzwanzig (as do all - other "_ZX" spoken numerals) Fix support for numbers in the - 10,000,000 to 99,999,999 range. Add support for numbers in the - 100,000,000 to 999,999,999 range. - -2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Remove an unused instance of an unnamed enum. - - * apps/app_meetme.c: Remove another duplicated definition - - * apps/app_meetme.c: Remove a variable that was declared twice. - - * codecs/gsm/Makefile: Add a couple more processors that need - optimizations excluded. (issue #8637) - - * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk. - AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same - thing. So, a digit would have been interpreted incorrectly here. - Since the channel driver will always have the begin and end - callbacks called for a digit, only support the button-down and - button-up messages. - - * .cleancount: Bump the cleancount since my last commit changed the - channel structure. - - * channels/chan_oss.c, main/rtp.c, main/channel.c, - channels/chan_phone.c, channels/chan_misdn.c, - channels/chan_skinny.c, channels/chan_features.c, - channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c, - channels/chan_zap.c, channels/chan_local.c, main/frame.c, - channels/chan_sip.c, channels/chan_agent.c, - include/asterisk/channel.h, channels/chan_gtalk.c, - channels/chan_iax2.c: Merge the changes from the - /team/group/vldtmf_fixup branch. The main bug being addressed - here is a problem introduced when two SIP channels using SIP INFO - dtmf have their media directly bridged. So, when a DTMF END frame - comes into Asterisk from an incoming INFO message, Asterisk would - try to emulate a digit of some length by first sending a DTMF - BEGIN frame and sending a DTMF END later timed off of incoming - audio. However, since there was no audio coming in, the DTMF_END - was never generated. This caused DTMF based features to no longer - work. To fix this, the core now knows when a channel doesn't care - about DTMF BEGIN frames (such as a SIP channel sending INFO - dtmf). If this is the case, then Asterisk will not emulate a - digit of some length, and will instead just pass through the - single DTMF END event. Channel drivers also now get passed the - length of the digit to their digit_end callback. This improves - SIP INFO support even further by enabling us to put the real - digit duration in the INFO message instead of a hard coded 250ms. - Also, for an incoming INFO message, the duration is read from the - frame and passed into the core instead of just getting ignored. - (issue #8597, maybe others...) - - * main/asterisk.c: Merged revisions 51300 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | - 4 lines Fix a memory leak on command line tab completion. The - container for the matches was freed, but the individual matches - themselves were not. (issue #8851, arkadia) ........ - -2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c: chan_zap compiles without libpri after - committing 7877 patch - - * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) - | 3 lines issue 7877: chan_zap module reload does not use - default/initialized values on subsequent loads. Reset - configuration variables to default values prior to parsing - configuration file. ........ - -2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com> - - * /: block this patch since it is already here - -2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, main/channel.c, main/pbx.c, - funcs/func_strings.c, main/app.c: Add some more checks for - option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, - patch(es) by tgrman - -2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com> - - * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in: - Ensure that the locations given to the Asterisk configure script - for ncurses, curses, termcap, or tinfo are further passed along - to the editline configure script. This fixes some - cross-compilation environments. (issue #8637, reported by ovi, - patch by me) - -2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 - Jan 2007) | 2 lines If a timezone is not specified, assume - localtime (instead of gmtime) (Issue #7748) ........ - -2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Only start timeout once we reach the end - of the files to play back. - -2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com> - - * main/cli.c: Fix an issue with file name completion in "module - load" and "load". Issue 8846 - -2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Copy MOH settings when calling a peer so - that if they put someone on hold or get put on hold themselves - they get the right music class. (issue #8840 reported by mdu113) - -2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com> - - * main/channel.c: Fix an issue with deprecated commands - -2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 - Jan 2007) | 2 lines Document all the fields, including the - indication that "uniqueid" should not be renamed. ........ - -2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com> - - * main/manager.c: Make the "hasmanager" option in users.conf - actually have an effect. (issue #8740, LnxPrgr3) - -2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Build the IMAP remote directory string - better and properly. Fix an issue with encoding the GSM voicemail - when attaching to the voicemail. (issue #8808 reported by - akohlsmith) - - * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames. - (issue #8840 reported by mdu113) - -2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com> - - * funcs/func_odbc.c: Fix some instances where when loading - func_odbc, a double-free could occur. Also, remove an unneeded - error message. If the failure condition is actually a memory - allocation failure, a log message will already be generated - automatically. - - * channels/chan_zap.c: Instead of dividing the offset by 2 - directly, make it more clear that the offset is being scaled by - the size of the elements in the buffer. (Inspired by a discussing - on the asterisk-dev list about this code) - - * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | - 3 lines Move the check for a failure of ast_channel_alloc() to - before locking the pvt structure again. Otherwise, on a failure, - this will cause a deadlock. ........ - -2007-01-17 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, main/utils.c: Merged revisions 51194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) - | 4 lines When ast_strip_quoted was called with a zero-length - string, it would treat a NULL as if it were the quoting character - (and would thus return the string in memory immediately following - the passed-in string). ........ - -2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: re-add "password" for realtime voicemail - -2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Return the correct result when directly writing out a - packet so that the core doesn't then decide to handle it the - regular way again. (issue #8833 reported by rcourtna) - -2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: a few more coding style cleanups and one - bug fix (from AnthonyL) - -2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the - scheduler callback. - -2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com> - - * main/rtp.c: Fix issue with dtmf continuation packets when the - dtmf digit is 0... Issue 8831 - - * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with - IMAP storage and realtime voicemail. Also update the vmdb sql - script for IMAP specific options. Issue 8819, initial patches by - bsmithurst (slightly modified by me) - - * doc/voicemail_odbc_postgresql.txt: change documentation to - reflect new procedure in 1.4/trunk - -2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions - 51161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) - | 2 lines Add documentation walkthrough on getting Postgres to - work with voicemail (from Issue 8513) ........ - - * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) - | 2 lines Postgres driver doesn't like a NULL pointer when - retrieving the length (Bug 8513) ........ - -2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: minor things i missed before i get jumped - on - -2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 51145 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 - lines Return previous behavior. ParkedCalls will be able to do - DTMF based transfers again. trunk however will get an option to - allow this to be set on/off. (issue #8804 reported by nortex) - ........ - -2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com> - - * main/file.c: Display more useful output when streaming files. - Include the channel name to which the file is being played. Issue - 8828, patch by junky. - -2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 - lines Add none as a valid callgroup/pickupgroup option. I - consider it a bug that it would inherit it all the way down and - not have any way to reset it to nothing - so that's why it is in - 1.2. (issue #8296 reported by gkloepfer) ........ - -2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com> - - * main/config.c: It is possible for the config pointer to be NULL - here, so it needs to be checked before dereferencing it. - -2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for - changing voicemail password in users.conf from voicemail main, - written by AnthonyL bug #8436 - -2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com> - - * Makefile.rules: Filter out a few CFLAGS that are not valid - CXXFLAGS. - -2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 - | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 - lines Solves issue with forwarding voicemails from folders other - than inbox. patch by anthonyl. ........ - -2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com> - - * main/asterisk.c: re-add deprecated "show version" CLI command. - -2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Move event processing into do_message so that it - gets executed again when events are tripped. - -2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com> - - * configure, include/asterisk/autoconfig.h.in, main/Makefile, - configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the - ACX_PTHREAD macro from the Autoconf macro archive for setting up - compiler pthreads support... should improve portability to - platforms with unusual pthreads requirements - -2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com> - - * main/astmm.c: Add missing newlines for two memory CLI commands. - -2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, - main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, - main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, - main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, - main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, - main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, - main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c, - main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, - main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, - main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, - main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, - main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, - main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c, - main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c, - main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, - main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 - Jan 2007) | 2 lines Bug 8814 - db should look for its header - using a relative path, instead of the system path (Fixes FreeWRT) - ........ - -2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, build_tools/make_sample_voicemail (added): when - building the sample greetings for maibox 1234@default during - 'make samples', build a greeting for each language and file - format the user selected to install with menuselect (reported by - Brian Capouch on asterisk-dev) - -2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Only write a frame out to the channel if one - exists. There are cases where one may not and would therefore - cause the channel driver to segfault. (issue #8434 reported by - slimey) - - * res/res_snmp.c: Only join the snmp thread on an unload if the - thread is actually running. (issue #8810 reported by junky) - -2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com> - - * configs/voicemail.conf.sample: Update documentation to state that - you shouldn't use realtime static with voicemail.conf - -2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com> - - * main/manager.c: We need to check for res being 0 in do_message - itself, otherwise our headers will get lost. - -2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com> - - * main/pbx.c, /: Merged revisions 50561 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) - | 2 lines minor documentation clarification ........ - -2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Remove check for channel state as it can - definitely be something other then ring, and also clean up the - code a bit. This should solve the parking issues and maybe some - attended transfer issues people have been seeing. - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add - support to see whether NAT was detected (yay symmetric RTP) and - also add a check in chan_sip so that if NAT has been detected and - the reinvite behind nat option has been turned off, then just do - partial bridge. (issue #8655 reported by mnicholson) - - * apps/app_speech_utils.c: Merge speech-multi branch which adds - support for joining multiple sound files together to be played - one after another in SpeechBackground. - - * main/config.c: Fix parsing when using something like ldap - settings. (done by anthonyl) - - * channels/chan_sip.c: Fix chan_sip not working issue. Let's not - prematurely return 0. (issue #8783 reported by st41ker) - -2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com> - - * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made - it fail to load if the config file existed. Issue 8777 - -2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 - lines Add another return value to dial_exec_full that indicates - execution is going to continuing at a new - extension/context/priority and to just let it slide. (issue #8598 - reported by jon) ........ - - * main/pbx.c: Ensure data's existence before trying to access it. - (issue #8774 reported by rcourtna) - -2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 50227 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | - 6 lines Make the number that represents the major version number - a single digit instead of 2. Using two digits makes it an octal - number when put into version.h, which breaks the compilation of - any out of tree module that checks the version for any version - after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev - mailing list, who gave credit to vihai for pointing it out) - ........ - -2007-01-09 17:11 +0000 [r50186] Jason Parker <jparker@digium.com> - - * main/cli.c: Re-add CLI command that should have only been - deprecated in 1.4. Thanks kshumard! (reported in person, so no - associated issue #) - -2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) - | 4 lines The advent of realtime has enabled people to use commas - in the fullname field. This could cause an issue with sending - voicemails, when the field is unquoted. (Issue 8595) ........ - -2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - handle re-invites properly in sip_hangup() - - Add some invitestate status changes just to be sure - -2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix an issue with voicemail and users.conf, - where it wouldn't ever parse a password, since it was using - "secret" instead of "password" Issue 8761, reported by and patch - suggestion from ssokol. - -2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com> - - * apps/app_senddtmf.c: we can't unlock a channel if we cant find - it. - AnthonyL bug #8741 - -2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Disable the more intense packet2packet bridging until - the bugs can be worked out. - -2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8677 - Handle failure of T.38 - re-invite This is not a fix, but adding an error message to tell - the admin that we have a bad configuration. We should not send - T.38 re-invites to devices that can't handle it (with the current - architecture where you have to hard-code t.38 support per - device). To really fix this, we need to figure out a way to tell - the incoming call that the re-invite failed, so we can signal - failure on that end and go back to the original call. - - * channels/chan_sip.c: Issue #8524, support multiple via header - values (tardieu) Thanks! - - * channels/chan_sip.c: We only need one forward declaration - - * channels/chan_sip.c: Issue 8735: Terminate state when extension - is unavailable for subscription - -2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 - lines Ensure we use the default refresh value of 60 if the remote - server does not send one. (issue #8746 reported by maethor) - ........ - -2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: since we use AC_PATH_TOOL to find tools, - we should use the results it provides for us (reported by Brian - Capouch on the asterisk-dev list) - -2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) - | 2 lines If openstream fails, then we crash (Issue 8564) - ........ - - * channels/chan_sip.c: Second condition was a subset of the first, - so hold was never decremented, thus hint stayed stuck (Issue - 8747) - -2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com> - - * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping - byte of allocated memory! This looks like it may have been a - chicken/egg scenario.. You had to call a cleanup func, because - everything was allocated. Then since you had to call a cleanup - func, you were forced to allocate - ie; strdup(""). - -2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com> - - * configure, acinclude.m4: one more time... - - * configure, acinclude.m4: proper fix for r49712 - - * configure, acinclude.m4: if --with-foo=<path> is specific for a - configure option, ensure that it is used for header file checking - as well - - * main/manager.c: ast_func_read() needs a writable copy of the - function name to be passed - -2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and - chan_zap also depend on zaptel. This fixes an issue (8727) with - zaptel being in a different directory, using --with-zaptel. - -2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c: don't 'consume' the params list before we try to - use it again - - * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c, - main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, - main/db.c, channels/chan_zap.c, channels/chan_sip.c, - apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, - utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c, - apps/app_queue.c, res/res_jabber.c: reduce stack consumption for - AMI and AMI/HTTP requests by nearly 20K in most cases - -2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Don't keep repeating the warning over and over - when the end of the call is reached. (issue #8724 reported by - xrg) - -2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_iax2.c: Merged revisions 49635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) - | 2 lines ensure that threads which are supposed to be detached - (because we aren't going to wait on them) are created properly - ........ - - * channels/chan_iax2.c: revert the dynamic_list insertion change... - that was not the right thing to do - - * channels/chan_iax2.c: create the IAX2 processing threads as - background threads so they will use smaller stacks when we create - a dynamic thread, put it on the dynamic_list right away so we - don't lose track of it - -2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: It's possible for the iax2 pvt to - disappear, so if it has... don't bother looking for dpentries. - -2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/threadstorage.h, main/asterisk.c, - build_tools/cflags.xml, include/asterisk.h, main/Makefile, - main/threadstorage.c (added), main/utils.c: add support for - tracking thread-local-storage objects that exist via - 'threadstorage' CLI commands - -2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com> - - * main/config.c: Only free comments and line buffer once we reach - the first level. (issue #8678 reported by ssokol, fixed by - anthonyl) - -2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com> - - * channels/iax2-parser.c, main/frame.c: don't mark these - allocations as 'cache' allocations when caching has been disabled - - * channels/iax2-parser.c: if we're going to decrement the frame - count when we free a frame, we should inrement it when we create - one :-) - - * channels/iax2-parser.c, channels/iax2-parser.h, - channels/chan_iax2.c: only do IAX2 frame caching for voice and - video frames - - * main/frame.c: don't do frame header caching in the core if - LOW_MEMORY is defined - - * channels/iax2-parser.c: don't define this type either if - LOW_MEMORY is enabled - -2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 - | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 - lines converted a lot of 256 to PATH_MAX and some white space - fixes. ........ - -2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com> - - * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode - - * codecs/Makefile: make building of codec_gsm against the system - GSM library actually work - -2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 - | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 - lines good catch russell sorry i missed that. fix magic number - with proper sizeof ........ - -2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com> - - * funcs/func_realtime.c: Fix the REALTIME() dialplan function. - ast_build_string() advances the string pointer to the position to - begin the next write into the buffer. So, this pointer can not be - used to copy the contents of the string later. The beginning of - the buffer must be saved. Interestingly enough, this code could - not have ever worked. (Pointed out by Sebb on IRC, thanks!) - -2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 - | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 - lines When using ODBC_STORAGE VoicemailMain doesn't create the - subdirectories for a mailbox such as the INBOX directory. this - patch solves that problem, was written by anthony be-125 ........ - -2007-01-03 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, - /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, - configs/misdn.conf.sample: Merged revisions - 48319,48321,48467,48552,48576,49135,49303 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | - 1 line changed a few debugs to higher debug levels ........ - r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | - 1 line added the export and import of the MISDN_ADDRESS_COMPLETE - Variable to inidcate wether the extension is already completely - dialed or if there might come additional digits by information - elements. also added some docs for that. ........ r48467 | - crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line - removed FIXUP state. added check for channel allocation conflict - when we create a setup while the other site creates a setup on - the same channel, besides the check we resolve this conflict. - ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 - Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a - preselected channel we just accept it, even when we're NT. added - some checks for segfaults. ........ r48576 | crichter | - 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we - reject a channel, because it's in use already, we shouldn't - process the setup anymore. made the channel allocation a bit - easier and more understandable, removed a few unused lines - ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 - Jan 2007) | 1 line added check for channel ranges in the - set/empty channel functions. set pmp_l1_check default to no. - added misdn restart pid cli command. added cleaning of channel - when we send a RELEASE_COMPLETE. ........ r49303 | crichter | - 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added - check for bridging in misdn_call to avoid setting - echocancellation when 2 mISDN channels are involved and when - bridging is set. That lead to a kernel panic before under - different situations, because we switched about 2 times between - hardware bridging and echocancelation * readded MISDN_URATE - variable which got lost before, this should make app_v110 work - again * fixed typo ........ - -2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, Makefile.rules: various Makefile improvements to get - chan_vpb (and any other C++ modules) to build properly - -2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Check pvt structure presence before passing - to send_command. This gets rid of the irritating message about a - packet without pvt structure. This happens because the scheduled - item is getting cancelled at almost the exact moment it is - getting executed. - -2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, - pbx/ael/ael.flex: This is a slight modification to Josh's edits - for #8579; both files edited were the produced by flex; so the - source files need to be changed instead, and the generated files - regenerated. - -2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always - enabled at that point in the code - -2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com> - - * main/pbx.c: Allow fractions of a second in the Wait() - application, like it says it allows. - -2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: remove comment that is unrelated to this - function - -2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net> - - * configs/features.conf.sample: Adding note on effect of - applicationmap features on re-invites - -2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, - configure.ac, codecs/codec_zap.c: check specifically for VLDTMF - and transcoding support in the system's Zaptel installation, and - make only the modules that need those features dependent on them - (this will allow building the other Zaptel-using parts of - Asterisk against older versions of Zaptel or those on other - platforms that haven't caught up yet to the Linux version) - - * Makefile: use a simpler (and portable) method to ensure that - menuselect is built as a host binary - - * Makefile: revert this change until a better solution can be - found... 'env -i' was not being used properly, but even when - changed to do so, this process fails during cross-compilation - because the menuselect build still sees 'CC' as set to the - cross-compiler - -2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: remove incomplete implementation of dnsmgr. - Let's fix this in trunk. - -2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_config.c: IAX has been deprecated for quite some time so - we had better use IAX2 when creating the dial string for users. - (issue #8697 reported by ssokol) - - * channels/chan_zap.c: Use asprintf to build the channel names - instead of custom function. I believe the custom function is - doing some things that are not portable across all - implementations. (issue #8570 reported by hterag & issue #8692 - reported by nicolasg) - - * main/rtp.c: If the Packet2Packet bridge is being broken because - of a masquerade then attempt to read a frame in so the masquerade - actually happens. Otherwise weirdness will occur. (issue #8696 - reported by kjotte) - - * channels/chan_iax2.c: Initialize the packet queue in load_module - instead of just declaring the list with the default value. (issue - #8695 reported by ssokol) - -2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have - comma args converted to vertical bars. I hope this change does - little harm. - -2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com> - - * /: put this value into the correct property - - * /, BUGS: Merged revisions 49045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) - | 2 lines location of the bug posting guidelines has changed - ........ - - * sample.call: simple commit to test CIA integration - -2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com> - - * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me - - * main/http.c: saw this in passing... fix a small typo - -2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: new versions of sounds - -2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com> - - * main/http.c: make the uris_lock a rwlock instead of a mutex lock - - needs to be forward ported to trunk - -2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/lock.h: Backport support for read/write locks. - -2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c, - pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, - pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, - pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk - from the ael stuff. Also, threw in a minor fix to frame.c to - avoid build-killing compiler warnings. - -2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com> - - * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not - available when LOW_MEMORY is used and things are being built in - the utils directory, so we need to resort to the old method of - strncpy. (issue #8579 reported by mottano) - -2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com> - - * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c, - main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, - main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c, - main/http.c, main/logger.c: since these variables all have static - duration, none of them need initializers (they default to zero - anyway) - - * include/asterisk/options.h, main/asterisk.c, main/file.c: move - extern declaration for this option to a header file where it - belongs provide an initial value for 'languageprefix' option, - instead of relying on randomness to provide a useful value - -2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Only include acl.h and lock.h once - - * channels/chan_sip.c: Only set rfc2833compensate flag once - (handle_common_options) - - * channels/chan_sip.c: - Remove checking for T38 options twice. - Keeping them in handle_common_options - -2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: make the option actually match the - documentation - - * channels/iax2-parser.c, include/asterisk/utils.h, - include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show - memory' and 'show memory summary' to distinguish memory - allocations that were done for caching purposes, so they don't - look like memory leaks - -2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more - politically correct - - * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy - cisco MWI support. Normally we try not to change our software for - bugs in other devices. But in this case, the Cisco phones are so - widespread so we try to implement a fix while waiting for a - bugfix from Cisco. - - * channels/chan_sip.c: - Make sure handle_common_options return 1 - when we found a common option - Move uncommon (only global) - option away from handle_common_options Reported by rizzo. Thanks! - - * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before - re-sending invite with auth. - - * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp. - (rizzo, #8600) - -2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Get rid of a needless memory allocation and - only create a conference structure in find_conf_realtime if data - was read from realtime. (issue #8669 reported by robl) - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an - API call that initializes an RTP structure. We need this because - chan_sip is cheeky and uses a temporary RTP structure for codec - purposes, and the API calls that are used rely on the lock. - (Pointed out on asterisk-dev by Andy Wang) - - * configure, configure.ac: Clean up autoconf file (gets rid of - warnings seen when rebuilding configure) and rebuild configure. - -2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com> - - * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | - 6 lines Fix an error introduced by copying and pasting the - handling of the >= operator for the MATH function. If a single - equal sign was used as an operator, the function would treat it - is as if it were the >= operator. Now, it properly handles it as - an invalid operator. (issue #8665, patch by tempest1) ........ - - * channels/chan_oss.c: Fix a typo in an error message that - indicated that the MGCP channel type could not be registered, - instead of the correct type, OSS. - - * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | - 3 lines Check for the proper return value on an error in a call - to mmap(). This was reported by Andy Wang on the asterisk-dev - list. Thanks! ........ - - * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | - 3 lines Remove a couple of misplaced dots in log messages. This - was reported by Andrea Spadaccini on the asterisk-dev mailing - list. ........ - - * main/http.c: Implement locking for the list of URI handlers to - make it thread-safe. - -2006-12-23 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0 released. - -2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com> - - * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. - - * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia - -2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com> - - * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 - lines Add new silence sound files to the spec for Redhat. (issue - #8652 reported by alvaro_palma_aste) ........ - -2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage - builds. - - * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so - it is then passed to the IMAP store file function. (issue #8614 - reported by punknow) - - * doc/snmp.txt: find is not the same as bind when it comes to - documentation. (issue #8626 reported by johann8384) - -2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com> - - * channels/Makefile: suppress compiler warnings in this module - until it can be improved - -2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 - lines Free localuser structure when we fail to dial (issue #8612 - reported by rizzo) ........ - -2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org> - - * apps/app_sms.c: fix a bogus datalen in the frames generated by - app_sms (causing noisy output if you listen to the output!) This - affects trunk as well, whereas 1.2 is ok. - -2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable - type for these unixODBC API calls, eliminating warnings on 64-bit - platforms that use the 'new' 64-bit types for ODBC API calls - -2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com> - - * Makefile: Use env -i to start a fresh environment when going to - build menuselect. This is more portable then using unset. (issue - #8543 reported by jtodd) - -2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/channel.h: unbreak the macro used for - incrementing the frame counters. I don't know when the bug was - introduced, but with the typical usage c->fin = - FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects - trunk as well (fix coming). - -2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Put thread into proper list if we abort - handling due to an error, and also hold the lock while putting it - back into the proper idle list so we don't prematurely get a - signal. (issue #8604 reported by arkadia) - -2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, - utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, - utils/ael_main.c: remove some now-unnecessary explicit includes - of autoconfig.h clean up per-file dependencies during 'make - clean' - - * build_tools/prep_tarball: need an additional argument here to - make the downloads actually occur - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep - these calls from thinking they have multiple arguments - - * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, - funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, - main, codecs/gsm, pbx, res, channels, codecs, utils, agi, - main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: - simplify dependency tracking system, using the compiler's - built-in method for generating them, and only doing dependency - tracking if developer mode is enabled via the configure script - - * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we - really, really have to have autoconfig.h included before all - other headers (especially system headers), the Makefile will now - force it to happen (this will fix build problems with files like - ast_expr2f.c, where we can't control the inclusion order in the - file itself) - - * funcs/func_curl.c: instead of initializing the curl library every - time the CURL() function is invoked, do it only once per thread - (this allows multiple calls to CURL() in the dialplan for a - channel to run much more quickly, and also to re-use connections - to the server) (thanks to JerJer for frequently complaining about - this performance problem) - -2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Turn payload_lock into bridge_lock and make it - encompass all RTP structure contents that may relate to bridge - information, including who we are bridged to. - - * channels/chan_iax2.c: Hold call structure lock in places where a - qualify or peer action can destroy it. - - * channels/chan_iax2.c: Lock network retransmission queue in all - places that it is used. - -2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported - from 1.2) - - * channels/chan_sip.c: Update to latest IANA spec - -2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Use a wakeup variable so that we don't wait - on IO indefinitely if packets need to be retransmitted. - - * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP - structure can change AFTER a bridge has started. This comes from - the packet handling of the SIP response when indication that it - was answered has been sent. Therefore we need to protect this - data with a lock when we read/write. (issue #8232 reported by - tgrman) - - * main/rtp.c: Remove direct RTCP bridging. I've come to the - conclusion that we should handle this through the core and not - just forward it on. Should solve a few bugs. - -2006-12-12 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta4 released. - -2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This - is the way it should have been done. - -2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com> - - * sounds/Makefile: new sounds package with 100% more silence - - * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.2 ........ - r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) - | 4 lines app_externalivr needs a real silence file, and - additional changes to add silence files into core instead of - extra patch provided by bug 8177 with minor additions. ........ - -2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Return non-existant callerid handling to - that which it was before. In 1.4 and trunk callerid can be - allocated but not have any contents so we have to use - ast_strlen_zero before passing it to the relevant functions. - (issue #8567 reported by pabelanger) - -2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: STRFTIME() does not actually require an - argument (issue 8540) - -2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Merge in my latest RTP changes. Break out RTP and - RTCP callback functions so they no longer share a common one. - - * apps/app_meetme.c: Use the correct API call to say a device state - changed. (Yes, I'm a nub.) - - * apps/app_meetme.c: Don't access the conference structure after it - has been freed. - -2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <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: Merged revisions 48374 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) - | 5 lines When doing a fork() and exec(), two problems existed - (Issue 8086): 1) Ignored signals stayed ignored after the exec(). - 2) Signals could possibly fire between the fork() and exec(), - causing Asterisk signal handlers within the child to execute, - which caused nasty race conditions. ........ - -2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 - line This version applies the patch suggested by stevens in bug - 7836 (make inbound channel RINGING state consistent with other - channels). ........ - -2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <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. In passing, go ahead and convert - this list to use the linked list macros. - -2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 - Dec 2006) | 3 lines Ensure that the file position is not - incremented beyond the total number of files available for - playback. (issue #8539, ulogic) ........ - -2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com> - - * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that - killed bug 8423 -- OriginateSuccess and OriginateError incomplete - channel name. May it rest in peace. - -2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being - retransmitted to Asterisk - -2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com> - - * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 - Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option - in the sample configuration file. (issue #8526, arkadia) ........ - -2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't send Contact on MESSAGE - -2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com> - - * configure.ac: Fix curl version number testing to be much more - friendly to non-bash shells. Issue 8508, patch by me. This - *SHOULD* be POSIX compliant now.. - -2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Merging the invitestate-1.4 branch after - successful testing. Will check if I can solve this with less - changes in 1.2. - - * configs/sip.conf.sample: Add missing s from another repository. - (thanks jcmoore!) - - * configs/sip.conf.sample: Updating sip.conf.sample with - information about T38 not working when chan_local or chan_agent - is involved in the call. I don't know how big a fix that would be - to solve, but this is the current state of affairs. (Chan_sip - currently checks if the other side of the bridge has a SIP tech. - We could/should implement another check, possibly for udptl_write - or some flag in the ast_channel structure). - -2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Oops, forgot to release the odbc handle - - * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) - | 6 lines If the recording in the database is too large, it will - fail to retrieve with an mmap error. Not too sure why this - doesn't happen when we put it in the database, also, but since - that doesn't seem to be broken, I'm not going to fix it (at least - until someone reports it). Solution is to ask for the file in - smaller chunks. (Bug 8385) ........ - -2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix an issue which didn't allow - unavail/greet/busy/etc messages from being saved into ODBC (and - probably IMAP). - -2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com> - - * configs/voicemail.conf.sample: Add documentation to - voicemail.conf.sample for ODBC storage. Issue 8499 - patch by - blitzrage. - - * doc/snmp.txt: Attempt to document some of the dependencies that - are needed for net-snmp Issue 8499 - initial patch by blitzrage. - -2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com> - - * sounds/Makefile: When "fetch" is in use, instead of "wget", - --continue is not a valid option. (issue #8451) - -2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Removing one of two pieces of code to - handle 481 response on INVITE - Move handling of REFER response - to handle_response_refer() - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, - configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax - transmission happens - Encapsulate RTP timers in the rtp - structure so we have one for video and one for audio The video - one is not used in 1.4, really. Will be used for RTP keepalives - when we can send something that video phones support in the RTP - stream. I now this is a big architectual change at this stage for - 1.4, but decided it was needed to avoid future bug reports. - - Document the RTP NAT keepalive option in sip.conf.sample Issue - 7679 in the bug tracker. Please test. - -2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h: Backport the comment containing the - warning regarding the limitations on the usage of this function. - It is thread safe, but not technically reentrant. - -2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) - | 2 lines if Dial() is going to send music-on-hold to the calling - party, it has to send PROGRESS first to ensure that the reverse - audio path has been setup first (BE-106) ........ - -2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com> - - * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: - FreeBSD 6.1 does not include wget by default. However, it has - fetch which will work just fine for our purposes of downloading - the sounds packages. So, check for both wget and fetch and the - configure script and use what was found to download them. If - neither one was found, and sound packages are selected that must - be downloaded, the install process will print out an informative - error message indicating the situation. Also, fix a couple places - where "make" was hard coded into some output messages by - replacing them with the $(MAKE) variable. (issue #8451, initial - patch by pabelanger, with additional modifications by me) - -2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 48183 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 - lines Fix a small typo - issue 8848, reported by pabelanger - ........ - -2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/cli.c: Double-unlock error (reported by blitzrage on IRC) - -2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the - "limitonpeers" patch from trunk, to fix a lot of issues with - queues and SIP device states - Remove support for T.38 early - media, since it's impossible. (Two patches in one - extra friday - evening offer due to being off line from svn today... :-) - -2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not - do a partial bridge for Google Talk since we need to handle STUN. - (issue #8448 reported by phsultan) - -2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue 8319 - change noncecount before - using it. - -2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 - lines Only print out debug message if bridged channel is not - NULL. (issue #8412 reported by jubilex) ........ - - * /, res/res_features.c: Merged revisions 48154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 - lines Do not listen for DTMF on the bridge that comes into - existence when ParkedCall is executed. This means native bridging - can now occur for this. (issue #8406 reported by kebl0155) - ........ - - * main/cdr.c, /: Merged revisions 48151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 - lines Print certain CDR messages out at the NOTICE level versus - WARNING since they can occur when used with the CDR applications - and are perfectly fine. (issue #8367 reported by dartvader) - ........ - - * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov - 2006) | 2 lines Document 'port' for SIP peers, came up because of - the current mailing list thread. (issue #8450 reported by - blitzrage) ........ - -2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net> - - * doc/manager.txt: Explain status reports and make codefreeze more - happy :-) - - * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by - GS 487 adapter without CSEQ on separate line in the REGISTER - request. Imported from 1.2. - -2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in - mm_login. (issue #8420 reported by slimey) - -2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Explain the use device status system - implemented in SIP for subscriptions, queues and manager a bit - better. Like in 1.2, you will get more detailed information if - you set a call limit for a device. When the call limit is - reached, the status system will report a device as busy. For - queues, setting a call limit per SIP device is propably a - requirement. In most cases, it will work much better if you only - use type=peer and not type=friend. We might decide to backport - the new setting from trunk to apply all call limits to the peer - part of a friend only. - -2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 48106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 - lines If the frame was duplicated before writing out then we need - to free it. (issue #8429 reported by edguy3) ........ - -2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. - -2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Don't crash if the mailstream was not - created. - -2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com> - - * Makefile: Export several more variables in top level Makefile. - Inspired by issue 8438. - -2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov - 2006) | 2 lines According to the research I have done we never - needed to include compiler.h in the first place so let's not! - (issue #8430 reported by edguy3) ........ - - * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 - lines Use the proper function to get the new message count - instead of always using the filesystem. (issue #8421 reported by - slimey) ........ - -2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 - Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) - ........ - -2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com> - - * main/manager.c: Remove a couple of unused variables (issue #8380, - casper) - -2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 - lines Do not reference the freed outgoing structure in the debug - message. (issue #8425 reported by arkadia) ........ - -2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change logging message - -2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com> - - * funcs/func_cdr.c: might as well also document the raw values of - the flag vars - - * /, funcs/func_cdr.c: A little bit of func_cdr documentation - upgrade-- no bug# involved, although 8221 may have inspired it. - -2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 - and future releases, you can disable subscription support totally - or per peer in sip.conf with allowsubscribe = yes | no - -2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com> - - * main/translate.c: bug 8189 posted this fix for main/translate.c - for PLC - -2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 - Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. - beatufied some logs, changed some loglevels. changed the default - value of block_on_alarm ........ - -2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't allocate unused variable. - -2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Video will never reach Packet2Packet bridging and can - do more harm then good. - -2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: If we have the non standard G726-32 setting turned on - we want to return G726-32 to the SDP, not our AAL2 string. (issue - #8330 reported by voipgate) - -2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 - provisional response. Let's not treat that as early media. - (discovered at the AVTF meeting in Paris). - -2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Oops, merge missed release of odbc object - - * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) - | 2 lines Failing to trap -1 error from mmap causes segfault - (Issue 8385) ........ - -2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com> - - * main/frame.c, /: Merged revisions 47859 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 - lines Don't forget to byte swap if we are exiting the smoother - feed early. (issue #8287 reported by arturs) ........ - -2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com> - - * /, doc/billing.txt: update documentation regarding IAX2 transfers - and CDRs Merged revisions 47776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) - | 2 lines update clearly wrong documentation regarding cdr_custom - ........ - -2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Compare technology using the pointers - instead of a straight comparison based on name. (issue #8228 - reported by dean bath) - -2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: check for pre-1.4 versions of Zaptel and - abort the configure script if found with an appropriate error - message - -2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD - notification optional, in order to avoid a lot of extra database - lookups for all those realtime users out there. - -2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47750 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov - 2006) | 2 lines Because of the way chan_local is written we - should be extra careful and make sure our callback functions have - a tech_pvt. (issue #8275 reported by mflorell) ........ - - * apps/app_meetme.c: Don't unreference the SLA object if there is - no SLA object in the devicestate callback. (issue #8354 reported - by loloski) - -2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup - - * UPGRADE.txt: Warn users about change in canreinvite - - * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never - authenticated (according to the RFC) - Update docs on - canreinvite. "nonat" is the recommended setting for most users - with phones behind a NAT. - -2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47711 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov - 2006) | 2 lines Make sure that the pvt structure exists before - trying to do fixup on Local channels. (issue #7937 reported by - mada123, fix by alamantia with mods by me) ........ - -2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL - -2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com> - - * main/channel.c: We need to ensure timelimit stuff is included as - well so warnings get played. (issue #8050 reported by KNK) - -2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com> - - * main/file.c: don't try to call fclose() if fopen() failed - -2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Improve SIP history - Never send reply to - ACK (again...) - -2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) - | 4 lines ensure that message duration is included in email - notifications for forwarded messages (BE-96, fix by me after - corydon used his clue-bat on me) ensure that duration in the - message metadata is updated if prepending is done during - forwarding (related to BE-96) remove prototype for API call that - does not exist ........ - - * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 - Nov 2006) | 2 lines clear the category's variable tail pointer as - well when variables are detached from it ........ r47688 | - kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 - lines when appending a list of variable to a category, ensure the - tail pointer points to the last variable in the list ........ - r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) - | 2 lines when re-writing the config file, don't repeat the path - if it hasn't changed ........ - - * main/config.c, /: Merged revisions 47682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) - | 2 lines ouch... don't use printf, use ast_log/ast_verbose - ........ - -2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: fix longest match search in find_cli. Trunk already - fixed. 1.2 not affected (well, i have no idea, the code is - totally different there). - -2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Send error message when we can't allocate - SIP dialog, possibly due to limitation of file descriptors. - (imported from 1.2) - -2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: If NAT detection is turned on or already detected - then say NAT is active when setting the remote RTP peer when - doing early bridging. (issue #8365 reported by marcelbarbulescu) - -2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com> - - * main/term.c: more formatting cleanup, and avoid running off the - end of the string - -2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Turn notice about unknown RTCP packet type into a - debug message instead. - -2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com> - - * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit - platforms (this variable is an 'int' anyway, comparing it to - 'signed long' is not useful) - -2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 - lines Update copyright information in the ADSI logo blob. - ........ - - * channels/chan_sip.c: Only keep the video RTP structure around if - 1. Video support is enabled and 2. A video codec is enabled on - the dialog - - * funcs/func_uri.c: Small documentation clarification for - URIENCODE. (issue #8294 reported by salaud) - -2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ - prefix did not completely transition app_voicemail when - ODBC_STORAGE is used (reported on IRC by caio1982, not in - bugtracker) - -2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com> - - * apps/app_amd.c: Use LOG_DEBUG to print out the indication that - app_amd is using default settings instead of using LOG_NOTICE. - This stops needless logging of this information under normal - circumstances. (issue #8361 reported by Seb7) - -2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Update documentation to fit the - implementation... - - * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in - retransmission system if it's an OPTION packet from peerpoke - -2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 - lines Initialize global pointers for connection and result to - NULL. (issue #8356 reported by james) ........ - -2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) - | 2 lines Having more than 255 old messages caused corruption in - the new/old count ........ - -2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com> - - * main/config.c: This solves bug 8342, whereby a crash occurs under - certain circumstances while reading a config file with comments-- - a call to CB_ADD shouldn't happen if withcomments is zero - -2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/cli.c, channels/chan_sip.c: Re-enable old deprecated - commands - -2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: - Don't reply to INVITE already replied - to when we get BYE - Declare errmsg as int. Oops. - -2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing - the messed if, but we all forgot to update the regressions. Until - now. - -2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being - found... just confuses users - -2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com> - - * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 - lines When sending an SMS with a user data header properly set - the UDH flag in the first byte. (issue #8347 reported by - hoffmeis) ........ - - * main/cli.c: Free full command string upon unregistering of CLI - command. Backported from revision 47536 from rizzo. - -2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Only produce error message about sip history - once - -2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com> - - * configure, acinclude.m4: AC_PROG_SED is included in autoconf - 2.60, but apparently it is not included in 2.59. So, to maintain - compatability with 2.59 since it is a small change, copy this - macro into acinclude.m4 and rename it to AST_PROG_SED. (issue - #8345) - -2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) - | 2 lines If the execute fails a second time, make sure that we - don't pass back a stale handle ........ - - * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) - | 2 lines Don't play dialtone if the seizing the channel fails - (Bug 7754) ........ - -2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks - DEA!!!) - - * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is - UDPTL in sdp... - - * channels/chan_sip.c: - Don't destroy SIP dialog because of a - failed T.38 re-invite. Wait for a bye. Final response to a - re-invite does not mean that the session dies, only that the - re-invite fails. - Keep RTP active during processing of T.38 - re-invite. If the re-invite fails, RTP needs to remain as before - the re-invite. Issue 8338 - darren1713. Please test. - - * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp - -Add some comments to t.38 code - -2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | - 4 lines Only do the check to determine whether the channel - calling this function is an IAX2 channel when getting the IP - address using the special argument, CURRENTCHANNEL. (issue #8341, - jcovert) ........ - - * Makefile: Add the target "menuconfig" as an alias for the - "menuselect" target. This is just a favor to users so that if you - accidentally type "make menuconfig" instead of "make menuselect", - it still works. (inspired by a comment on IRC from wangster - calling me an "especially devious asterisk developer" for having - it be menuselect instead of menuconfig. :) ) - - * main/term.c: Tweak the formatting of this new function to better - conform to coding guidelines. - -2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com> - - * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo - safe output! - -2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we don't use 32 bits when we only - need one bit. - -2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: ...and make sure that the dialog is - destroyed, even if we don't get any answer on the bye... This is - the channel that remains dead after the SIP transfer - - * channels/chan_sip.c: Add debug output while trying to trace bug - in bug report - - * channels/chan_sip.c: Make sure we destroy dialog... - - * /, channels/chan_sip.c: Small cleanup of handle_request_invite() - - imported from 1.2 with changes - -2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide - callerid name for switches that bork on it. - -2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart - SDP (alphaque) - -2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org> - - * build_tools/prep_moduledeps: grep -m is not available on BSD, so - use head -1 instead - -2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c: Only split up extension and context if a - value exists. (issue #8332 reported by loloski) - -2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_iax2.c: Discussion of these CLI changes resulted in - more consistency (Bug 8236) - -2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then - removing them should be LOG_NOTICE, not LOG_DEBUG - - * apps/app_queue.c: reflect addition/removal of dynamic queue - members in queue_log, so that people using dialplan replacement - for AgentCallbackLogin can still track login/logout (issue #7736, - reported/patched by whoiswes but this commit was written by me - and covers all three paths for AQM/RQM) - -2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Rip out half implementation of 491 response - support, since it wasn't implemented properly and caused memory - leaks in the case of us getting 491's, which Asterisk actually - sends... Since it is a bit too complicated to fix this, I'll rip - it out of 1.4 and put it on the to-do-list for future releases. - Now, we handle this as congestion, which it really is. Issue - #8331 - - * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. - Thanks fenlander! - -2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Fix building of chan_h323 by completeing - some structure definitions. (issue #8327 reported by Mithraen) - - * apps/app_voicemail.c: Do conversion in a more easier to read and - working way for \r, \n, and \t. (issue #8324 reported by - johnlange) - -2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, channels/chan_zap.c, - build_tools/prep_moduledeps: Work around an issue that caused - menuselect to display a bogus description for app_voicemail and - chan_zap. These modules use some preprocessor directives to - determine what it will report to Asterisk as its description. - However, the way we extract this information from the source - files for menuselect is not smart enough to figure this out. - (issue #8326, #8328) - -2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov - 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and - higher as, well, it's apparently going to be removed. This should - make all you FC6 fans happy as your Asterisk will now build - without any mods. ........ - -2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com> - - * main/cli.c: fix tab completion for "core debug channel" and "core - no debug channel" - - * main/cli.c: Fix "core show channel". Also, fix tab completion for - both "core show channel" and "core show channels". - - * main/cli.c: Fix "core debug channel <whatever>". I guess someone - needs to go through and audit every CLI command that changed - number of arguments ... - - * main/asterisk.c: revert the previous change, which actually - modified the deprecated command, "show profile". Now, actually - apply the change to "core show profile". - - * main/asterisk.c: Fix argument parsing for the "core show profile" - CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) - - * main/cli.c: Fix another CLI command, "core show uptime" ... - (issue #8323, reported by johnlange, fixed by myself) - - * main/asterisk.c: fix "core show version" to reflect the new - number of arguments for this CLI command (issue #8316, kshumard) - -2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com> - - * main/channel.c: This update fixes 7531 - - * channels/chan_skinny.c: Committed in behalf of 8190. - -2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c: the battle over CLI command formats has broken - stuff... - - * channels/chan_sip.c: add simple fix for SDP to report proper - sample rate for G.722 media sessions - -2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com> - - * utils/streamplayer.c: I occasionally get email from users that - are trying to figure out what this does, or due to some - misunderstanding as to what it is supposed to do, can't get it to - work. So, I have added some text here to hopefully explain what - this application does and does not do. - - * channels/chan_gtalk.c: Make this module build again - - * configure, configure.ac, acinclude.m4: Copy the macros from - libtool.m4 to our own acinclude.m4 such that libtool is no longer - required to be installed to be able to generated the configure - script. - -2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) - -2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com> - - * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, - include/asterisk/stringfields.h, apps/app_voicemail.c, - main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, - channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, - channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, - channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to - solve the problem in bug 7506. It's a lot of rework to solve a - fairly small problem... such is life. - -2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: Make MOH work as it did before in - chan_local, without this then it can go funky when transfers and - MOH are involved. (issue #7671 reported by jmls) - -2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com> - - * configs/musiconhold.conf.sample: clean up sample config, and make - native file playback the more obvious default choice - -2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: large overhaul to voicemail imap support. - Allows support for more imap servers, also a better - implementation of several parts of the original work. patch - provided by 8033 with major upgrades. - -2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of - continue. - -2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Fixing the attack shield so it doesn't - produce attacks... Issue 8265 - never reply to an ACK - -2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 - Nov 2006) | 5 lines If random order is enabled for files mode - music on hold, set a random initial position, instead of always - starting at the first file, and doing the random operation only - when switching to the next file. (bug reported by John Lange on - the asterisk-dev mailing list) ........ - -2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and - transfer from "john" Thank you! - -2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com> - - * main/cli.c: Fix another bug in "core set debug" ... - - * main/asterisk.c, main/cli.c: Really fix the "core set debug" and - "core set verbose" CLI commands. - - * main/cli.c: fix the "atleast" option to the "core set verbose" - and "core set debug" CLI commands - -2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: This fix introduced via bug 8233 - -2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org> - - * bootstrap.sh: align bootstrap.sh with the version in trunk (needs - to be blocked as it is already in trunk) - - * configure.ac: add proper environment vars to detect modules on - freebsd. (already applied to trunk so it needs to be blocked - there) - -2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, - channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More - changes making the CLI more consistent with "category verb - arguments" (continuation of issue 8236) - - * main/config.c, main/cli.c, main/channel.c, main/manager.c, - channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, - main/http.c, main/file.c, main/logger.c, main/image.c, - res/res_indications.c, main/asterisk.c, res/res_odbc.c, - channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, - channels/chan_local.c, main/frame.c, channels/chan_sip.c, - res/res_features.c, channels/chan_agent.c, res/res_crypto.c, - res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: - Reverse change of "show" to "list" and make several other - commands more consistent with "category verb arguments" - -2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Move check for codec translation to - sip_call() instead of in add_sdp. No one bothers with the result - of add_sdp anyway... Yet... - - * channels/chan_sip.c: Disable code for T38 over TCP and RTP since - there's no trace of actual functionality for it :-) - -2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 - Nov 2006) | 3 lines ignore files in a music on hold directory - that begin with '.' (issue #8249, cboie) ........ - -2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix - -2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't send INVITE when we have determined - that we can't offer any audio formats due to lack of transcoding - support (or incorrect configuration) - -2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 - lines Repeat after me oej: I will at least make sure my code - compiles before I commit it. ........ - -2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) - -2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com> - - * /, main/callerid.c: Add the missing call to free described in - issue #8268. Also, add a bunch of missing calls to free in - callerid_feed_jp(). - - * main/say.c: fix saying one hundred and two hundred in hebrew - (issue #7810, eldadran) - - * Makefile, configure, codecs/gsm/Makefile, configure.ac, - build_tools/strip_nonapi, makeopts.in: Fixes for - cross-compilation on mips (issue #8058, ywalther, with some - modifications) - - * aclocal.m4, build_tools/menuselect-deps.in, configure, - build_tools/embed_modules.xml, configure.ac: Add a check in the - configure script to determine whether ld is GNU ld or not. This - is needed because module embedding only works for gnu ld. GNU ld - is now listed as a dependency for all of the module embedding - options in menuselect. (issue #8143) - -2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_gtalk.c: bind address support from bug 8164 - -2006-11-01 19:49 +0000 [r46802] 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 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com> - - * main/Makefile: Force poll() emulation for Darwin to always be on. - It's too broken to consider being used. This resolves the console - issue OSX users have been seeing. I would have liked to autoconf - this but I haven't been able to come up with a test case that - works. Que sera. - -2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com> - - * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | - 9 lines soxmix and Asterisk expect different file extensions for - certain formats. This was already handled for the wav49 format. - However, it was not handled for ulaw and alaw. I fixed this in - such a way that using the alternate extensions for ulaw and alaw - will only happen if we know we're calling soxmix, and not a - custom script defined using the MONITOR_EXEC variable. The wav49 - processing was left alone so that external scripts will see no - behavior change. (issue #7550, reported by mnicholson, proposed - patch by junky, committed fix is a bit different) ........ - -2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: It's another round of chan_iax2 fixes! - Should hopefully fix the deadlock issues people have been - reporting. IAXtel now has qualify turned on for 800 peers and it - is handling it fine. - -2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com> - - * main/config.c: Cleanups suggested by Russell. - -2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Prevent an infinite loop when config - processing gets to a jitterbuffer option - -2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com> - - * main/translate.c: Fix "core show translation" output. Issue - #8243, patch by Damin. - -2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/translate.h, main/translate.c: add an API so - that translators can activate/deactivate themselves when needed - - * include/asterisk/translate.h, main/translate.c: revert changes - that were the wrong way to address this... proper fix coming - - * main/translate.c: let's set the seen flag early enough to - actually make a difference... - - * include/asterisk/translate.h, main/translate.c: don't re-do setup - operations for translators that can dynamically register - themselves - -2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net> - - * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue - #8089 - Fix the ENUM support (picking one record by number). - Thanks otmar! - - * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport - when we're supposed to support ;rport. Issue #7473. - - * /, channels/chan_sip.c: If peer fails ACL check, fail peer at - REGISTER - - * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! - -2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com> - - * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the - boot process to ensure it starts after stuff like MySQL (issue - #8253, Alric) - - * /, main/utils.c: Merged revisions 46560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | - 3 lines When handling the case where the hostname is just an IPV4 - numeric address, be sure to set the address type. (issue #8247, - alexr) ........ - - * /, res/res_agi.c: Merged revisions 46557 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | - 3 lines fix some copy/paste bugs in the checking of arguments for - the "control stream file" AGI command (issue #8255, mnicholson) - ........ - - * main/translate.c: Add a small tweak to the code that checks to - see whether destination formats are translatable based on the - source format. If we have already determined that there is no - translation path in one direction, don't bother checking the - other direction. - -2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com> - - * main/translate.c: when unregistering a translator, don't rebuild - the translation matrix unless needed when filtering formats out - of an offer, ensure we check for translation ability in both - directions - - * include/asterisk/linkedlists.h: ensure that items removed from a - list are always unlinked from the list (next pointer set to NULL) - -2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Don't explicitly link in crypt as it is - not used on some platforms. - - * channels/chan_iax2.c: We need to lock the pvt structure during - retransmission as another worker thread may be doing something as - well. - -2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net> - - * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, - include/asterisk/doxyref.h, channels/chan_sip.c, - main/ast_expr2f.c, include/asterisk/module.h, - formats/format_ogg_vorbis.c, main/app.c, - include/asterisk/channel.h, include/asterisk/lock.h, - include/asterisk/frame.h: Issue #8246 - Doxygen fixes from - kshumard. An extra big thankyou is given to everyone that - contributes to doxygen! THANK YOU! - - * main/rtp.c, /: Bind RTCP to the same IP as RTP - - * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 - redirects (imported from 1.2) - - * /, channels/chan_sip.c: Issue #7608 - Notifications sent with - wrong content-type (imported from 1.2, modified) - - * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that - was reported for trunk, but obviously exists in 1.4 too. - - * channels/chan_sip.c: Restoring the old logic, since working - around it and fixing it seemed too complicated. - The - SIP_OUTGOING flag indicates the direction of the last transaction - in the dialog. - The initreq stores the last request in the - dialog, the request that opened the latest transaction. Please - now retry all the 1.4 bug reports with mixed to/from headers, - tags etc in ACK, BYE, CANCEL. Thanks! - - * channels/chan_sip.c: Accepting a message twice may be - misinterpreted... - - * channels/chan_sip.c: - 183 is not reliable message... - Error - should not have SDP - -2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com> - - * utils/Makefile: Don't build muted on OpenBSD, it is not - supported. - -2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: move the copy of the default settings to the - global settings back out of process_zap, so that they aren't - overwritten when process_zap is called multiple times - -2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net> - - * contrib/asterisk-ng-doxygen: Put some doxygen pressure on - Christian :-) - -2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com> - - * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, - res/res_musiconhold.c: We should always be using _exit() after a - fork() or vfork() instead of exit(). This is because exit() does - some extra cleanup which in some implementations of vfork(), for - example, can actually modify the state of the parent process, - causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) - - * channels/chan_zap.c: Instead of iterating all of the options once - to look for jitterbuffer options, and then again for everything - else, move the processing of jitterbuffer options into the main - loop so that there are no erroneous messages about ignoring - unknown options. (issue #8226) - -2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 46350 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | - 1 line fixed a bug which caused chan_misdn to try to allocate 2 - times the same channel on high load, which then caused - instability of mISDN. removed a useless function from isdn_lib.c - ........ - - * channels/misdn_config.c: fixed not compile issue, which was just - introduced - - * channels/misdn_config.c, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - Merged revisions 46176 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | - 1 line added nttimeout option to configure wether we disconnect - calls on NT timeouts or not during an overlapdial session - ........ - -2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com> - - * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 - lines oops - somebody forgot to change this - long ago, probably. - ........ - - * CHANGES: grammar check - -2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net> - - * CHANGES: Corrections to changes (Multiparking is not included) - -2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com> - - * main/translate.c: - If the source has no audio or no video - portion, do not call powerof() to get the format index. - Don't - run through the audio and video loops if there is no audio or - video portion of the source If 0 is passed to powerof, it will - return -1. This value of -1 was then being used as an array index - in these loops, which caused a crash on some systems. Other than - this issue, this code works as we expected it to. If a format is - not in the source, and we have to translation path to it, it is - not offered in the list of acceptable destination formats. (fixes - issue #8231) - -2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com> - - * CHANGES: update to reflect G.722 addition - -2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com> - - * doc/backtrace.txt: update backtrace documentation to reflect - changes in 1.4 (issue #8230, kshumard) - -2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com> - - * main/config.c, main/manager.c: Fix config comment code - preservation code (thanks murf!) - -2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Old todo note - Don't add Contact header on - BYE and Cancel - -2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com> - - * configure.ac: fix error output when checking for openh323 to - refer to openh323 instead of pwlib (issue #8222, misaksen) - -2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Somewhat ugly code to try to fix issue - #7608. Since the problem was not very well defined, the fix is a - bit fuzzy too... Thanks to Luigi for accidentally spotting the - possible problem! - -2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: update warning message to include "agi" option - (issue #8225, jmls) - -2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: use 1.4.3 extra sounds with corrected silence - files - - * sounds/sounds.xml, sounds/Makefile: add support for prebuilt - G.722 prompts and music on hold files - -2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: show settings doesn't produce a list of - similar objects, it should stay a "show" - -2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com> - - * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, - channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, - pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, - main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, - cdr/cdr_custom.c, channels/chan_mgcp.c, - apps/app_parkandannounce.c, apps/app_voicemail.c, - channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, - res/res_adsi.c, main/utils.c, apps/app_ices.c, - pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, - apps/app_getcpeid.c: apparently developers are still not aware - that they should be use ast_copy_string instead of strncpy... fix - up many more users, and fix some bugs in the process - -2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/pbx.c: WaitExten truncates decimals of times to wait, - instead of accepting them (Bug 8208) - -2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com> - - * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, - channels/chan_h323.c, channels/chan_iax2.c, - include/asterisk/frame.h: add passthrough and file format support - for G.722 16KHz audio (issue #5084, original patch by andrew, - updated by mithraen) - - * channels/chan_sip.c, main/translate.c: code zone experiment: - don't offer formats in the outbound INVITE that aren't either - passthrough or translatable - - * main/translate.c: if multiple translators are registered for the - same source/dest combination, ensure that the lowest-cost one is - always inserted earlier in the list - -2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com> - - * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, - #8147) - -2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: We need to initialize our scheduler pthread - condition... yes. - -2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org> - - * main/http.c: merge 45152 don't leak descriptors in http.c - - * channels/chan_sip.c: merge 45966 refer_to_domain potentially - containing options - - * channels/chan_sip.c: merge 46026 improper checks on get_header() - return values - - * channels/chan_sip.c: merge 46045 prevent NULL args to - ast_strdupa() in chan_sip.c - -2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com> - - * Makefile: Restore the ability to remove the firmware directory - without causing the installation to fail (issue #8111) - -2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com> - - * main/translate.c: ensure that the translation matrix is properly - lock-protected every place it is used - - * include/asterisk/translate.h, main/translate.c: add an API call - to allow channel drivers to determine which media formats are - compatible (passthrough or transcode) with the format an existing - channel is already using - - * doc/imapstorage.txt: simplify and correct voicemail IMAP storage - build instructions - -2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/channel.c: Pass through a frame if we don't know what it is, - rather than trying to pass a NULL, which will segfault a channel - driver (Bug 8149) - -2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com> - - * utils/muted.c, utils/ael_main.c: In muted.c, check the return - value of strdup. In ael_main.c, check the return value of calloc. - (issue #8157) In passing fix a few minor bugs in ael_main.c. The - last argument to strncpy() was a hard-coded 100, where it should - have been 99. I changed this to use sizeof() - 1. - - * apps/app_meetme.c: Fix the descriptions of some of the - MeetMeAdmin options (issue #8098, mflorell) - - * res/res_jabber.c: don't crash when an incoming message has no - "from" (issue #8205, jmls) - -2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 - lines Don't leak memory mmmk? ........ - -2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 - Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and - couldn't be initialized it would cause a segfault after 'reload'. - Reported by Drew/Matt thx. ........ - -2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com> - - * res/res_monitor.c: Add a couple missing unregistrations of - manager actions and remove duplicate unregistrations of - applications. (issue #8194, jmls) - -2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com> - - * main/loader.c: Don't use promotion on Darwin because it doesn't - seem to work quite right in all cases, this should solve the - unresolved symbol issue people have been seeing. - - * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get - installed in the proper location (reported on asterisk-dev - mailing list) - -2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Let's understand SIP: - REFER can create - dialog, Asterisk does not support it yet - NOTIFY can create - dialog in Asterisk's implementation (voicemail) even though we - don't support the server side of it. In this case, the standard - is a side issue ;-) - Added extened functionality for unsupported - methods (PING, PUBLISH) so we don't create PVT's for those - either. Russellb needs to judge what to do with this in 1.2, but - I think the current implementation n 1.2 is a bug since we're - sending bad replies to NOTIFY and REFER outside of dialogs - -2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com> - - * res/res_jabber.c: Let's remember to unregister JabberStatus too - (issue #8184 reported by jmls) - - * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct - 2006) | 2 lines Respect language selection when seeing if the - file exists (issue #8178 reported by mnicholson) ........ - - * channels/chan_sip.c: If the jitterbuffer is forced on then we - can't partially bridge (reported by wangster on #asterisk-dev) - -2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Don't leak the actual thread-specific - sip_pvt struct - -2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't leak memory when a chan_sip thread is - destroyed that has a thread-local temp_pvt allocated - -2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Don't modify things if we are using vfork as - this is very bad and may cause unexpected behavior (issue #7970 - reported by Nick Gavrikov) - -2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: remove duplicate declarations - -2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org> - - * main/http.c: merge from trunk: move ast_variables_destroy() to a - better place in handle_uri() to avoid leaking memory on non - existing files. - -2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Don't segfault if you're using a channel driver that - doesn't turn RTCP on - -2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com> - - * main/channel.c: Don't attempt to access private data members of - the pthread_mutex_t object, because this does not work on all - linux systems. Instead, just access the reentrancy field in the - ast_mutex_info struct when DEBUG_THREADS is enabled. If - DEBUG_CHANNEL_LOCKS is enabled, the developer probably has - DEBUG_THREADS on as well. (issue #8139, me) - - * configs/sip_notify.conf.sample: update entry to reboot a snom - phone (issue #7850, pnlarsson) - -2006-10-17 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta3 released. - -2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/stringfields.h, main/ast_expr2.c, - main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: - optimize the 'quick response' code a bit more... no more malloc() - or memset() for each response expand stringfields API a bit to - allow reusing the stringfield pool on a structure when needed, - and remove some unnecessary code when the structure was being - freed - -2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <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 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Adding information about Marks - direct-RTP hack to the docs... - -2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com> - - * LICENSE: provide licensing language for IAXy firmware file - -2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new - directed pickup (BE-85). - -2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net> - - * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for - your support! - - * channels/chan_sip.c: Don't destroy dialog for unexpected REFER - response... - -2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com> - - * funcs/func_rand.c: update the doc string for both AEL and - extensions.conf users. - -2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com> - - * main/acl.c don't drop the entire permit/deny list when an attempt - is made to add an invalid entry (BE-92) - -2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c: Clear the quiet flag too since we are - restarting a recognition again (reported on -dev by Stephan - Edelman) - - * res/res_speech.c: Check return value from engine in case of - failure (ie: out of licenses) (reported on -dev mailing list) - -2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest17 (added), - pbx/ael/ael-test/ael-vtest17/extensions.ael (added), - pbx/ael/ael-test/ael-vtest17 (added), - pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in - this release via these changes - -2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: avoiding warning, fixing potential bug - -2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com> - - * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, - codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, - codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, - codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, - codecs/lpc10/synths.c, codecs/lpc10/preemp.c, - codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, - codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, - codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, - codecs/lpc10/lpcini.c, codecs/lpc10/random.c, - codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, - codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, - codecs/lpc10/analys.c, codecs/lpc10/onset.c, - codecs/lpc10/energy.c, codecs/lpc10/deemp.c, - codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, - codecs/lpc10/median.c, codecs/lpc10/encode.c, - codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, - codecs/lpc10/invert.c: And file said... let the compiler warnings - STOP! - - * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 - reported by mnicholson) - - * apps/app_playback.c: Move say.conf existence check to do_say - function since it is called from multiple places (issue #8144 - reported by kshumard) - -2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <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 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Complete merging in RPID screen changes - (issue #8101 reported by hristo, patch by oej in revision 44757) - - * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add - the background refresh item back into the scheduler if enabled - since it is deleted during reload. (issue #8142 reported by - p_lindheimer) - -2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/utils.c: use a configure script test for PMTU discovery - control instead of just assuming it's available on Linux - -2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some - echocandisable issues when bridged. this caused a kernel panic - sometimes.. also some minor formatting fixes - - * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause - got a wrong isdn cause at RELEASE_COMPLETE - -2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: merge formatting and minor code - simplifications from trunk - -2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_gtalk.c: fix for bug 7764. - -2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: we can only send one 'a=ptime' attribute per - media session, not one for each format - - * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, - main/utils.c: ensure that IAX2 and SIP sockets allow UDP - fragmentation when running on Linux (thanks to Brian Candler on - the asterisk-dev list for the tip) - -2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com> - - * main/manager.c: fix a silly typo in a comment that I saw while - reading the commit list - -2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com> - - * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue - #8135 reported by ssokol) - -2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com> - - * main/manager.c: append_event must be called while holding the - session lock - -2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: change some debug output to use LOG_DEBUG - instead of verbose output - -2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com> - - * main/db1-ast/Makefile: These are already set by the parent - Makefile.. There is no need to have this here (it doesn't - actually work anyways). - -2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: removed warning because of missing - prototype declaration - -2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Do not set default/global values in the - variable declaration, set it in reload_config() - -2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Move some stuff around so that a NOTIFY - dialog won't hang around until the end of the world under certain - circumstances - -2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz> - - * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: - CHANNEL() function sometime mix parameter and value - -2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_logic.c: Lost of a bit of logic when this was - simplified between 1.2 and 1.4 (Bug 8117) - -2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Bail out if we have no refer structure and - we get a refer response - -2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: more merge from trunk (comments and change a - static function name) - -2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only set DTMF information if an RTP - structure exists - -2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added - support of dynamically enabling hdlc on bchannels - -2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: whitespace changes related to previous - commit - - * channels/chan_sip.c: merge a few code simplifications that have - gone into trunk during last week, to reduce differences between - the two branches and make porting fixes easier. - -2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix a problem where phones that go - "missing" never got unregistered. Issue #8067, reported by pj, - patch by Anthony LaMantia (with minor whitespace modifications) - -2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid - the deadlock - - * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup - (issue #8115 reported by vazir) - -2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: do not dereference p if we - know it is NULL - -2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate - caller's transfer capability too - -2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: put common code in a - function to avoid repetitions. - - * channels/chan_sip.c: remove hardwired usage of 5060, use - DEFAULT_SIP_PORT instead - - * channels/chan_sip.c: option_debug checking - before printing to debug channel. - - * channels/chan_sip.c: backport simplifications on sip_register, - usage of ast_set2_flag(), and fixes to the handling of failed - module loading. - - * channels/chan_sip.c: improve and document function - get_in_brackets(), introducing a helper function - find_closing_quote() of more general use. - -2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/linkedlists.h: ensure that mutex locks inside - list heads are initialized properly on platforms that require - constructor initialization (issue #8029, patch from timrobbins) - - * CHANGES: remove Jingle as per mog - -2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Remove the seqno check for RFC2833, the handler is - smart enough to not need it. - -2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com> - - * CHANGES: various cleanups - -2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: When the sequence number rolls over then reset the - recorded sequence number for DTMF (issue #8106 reported by - bungalow) - - * main/file.c: Even more frames to treat as though the remote side - disappeared (issue #8097 reported by eldadran) - -2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c, main/http.c: make sure sockets are blocking when - they should be blocking. - -2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed segfault which happens during - hold/transfer action - - * channels/chan_misdn.c: if INFORMATION Message come with keypad - instead of called party number, we just use the keypad as called - party number. - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - added the option 'reject_cause' to make it possible to set - the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, - which is automatically rejected because chan_misdn does not - support that kind of callwaiting. Therefore chan_misdn supports - now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc - now gets the info if the requested channel is incoming or - outgoing to make the 3. channel possible - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, - removed a useless bc field, added setting of frame.delivery fields, - some minor code cleanups - -2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com> - - * main/file.c: Treat busy control frames as hangup in the file streaming - core (issue #8097 reported by eldadran) - -2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. - Many thanks to Doug! - -2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite - hanging by a thread if the other side is already setup with T.38 - -2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com> - - * main/app.c: don't segfault when an argument without a close - parenthesis is found stop parsing as soon as that situation - occurs - -2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com> - - * CHANGES: I put the accumulated changes from the commit logs and - inspection, into CHANGES. Hope everyone approves! - - * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the - install process sticks muted.conf in /etc/asterisk, so that's - where muted should look for it, right? - -2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't totally bail out if T.38 was - negotiated - -2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix Polycom presence notification again - -2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org> - - * utils/Makefile: as far as i can tell astman only uses newt... - - * Makefile: put linker flags in ASTLDFLAGS where they belong - -2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE - requests add workaround for new Polycom firmware SUBSCRIBE - requests (bug is known to exist in 2.0.1 firmware) - - * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually - work - -2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, - pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, - pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, - pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, - pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, - pbx/ael/ael-test/ael-test16/extensions.ael (added), - pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, - pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, - pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, - pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the - problems reported in bug 8090 - -2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, - main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, - channels/chan_skinny.c, channels/chan_h323.c, main/http.c, - channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, - main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, - include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, - channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, - main/devicestate.c, main/utils.c, res/res_musiconhold.c, - channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update - thread creation code a bit reduce standard thread stack size - slightly to allow the pthreads library to allocate the stack+data - and not overflow a power-of-2 allocation in the kernel and waste - memory/address space add a new stack size for 'background' - threads (those that don't handle PBX calls) when LOW_MEMORY is - defined - -2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com> - - * configs/muted.conf.sample: I've been meaning to add some - explanation about muted... here it is - - * configs/manager.conf.sample: CLI reverbification update to this - config file - - * apps/app_macro.c: In response to bug 7776, a Warning has been - added to the doc string for Macro(). - -2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com> - - * main/asterisk.c, main/loader.c, main/term.c, Makefile, - include/asterisk.h: ensure that local include files are always - used avoid a duplicate function name (term_init()) - -2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing - client without resource. - -2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix a logic error in my previous fix to the queue - reload code - -2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Change default presentation indicator - to "user provided not screened" if octet 3a missed in - CallingPartyNumber IE - -2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Use VideoSupport instead so it is considered - a valid XML attribute name. (issue #8075 reported by renemendoza) - -2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Fix preparation of type and - presentation of calling number - -2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com> - - * doc/jingle.txt, channels/chan_jingle.c (removed), - include/asterisk/jabber.h, configs/jingle.conf.sample (removed), - res/res_jabber.c: updated res_jabber for even better component - support, soon will be jep-0100 compliant. also removed - chan_jingle and infromed info from jingle.txt, chan_gtalk still - works and should be used in this version. - -2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <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) - -2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz> - - * configs/users.conf.sample, pbx/pbx_config.c: Missed part of - userconf functionality for chan_h323 - -2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com> - - * main/io.c: Shrink when current_ioc is unused. It is set to -1 when - unused, not 0. (issue #7941 reported by eclubb) - -2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz> - - * doc/realtime.txt: Typo fix - - * channels/chan_h323.c: Optimization of oh323_indicate(): less - locks - less problems, plus single exit point - -2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when - you're not talking about a channel :) - -2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Do not simulate any audio tones if we got - PROGRESS message - -2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com> - - * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to - be empty. The cause is that since ASTDATADIR is explicitly - exported using "export ASTDATADIR" at the top of the Makefile, - make no longer considers the variable "undefined", so the - Makefile can't use ?= to set ASTDATADIR if not yet set. (issue - #8063, reported by akohlsmith, fixed by me) - - * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" - option in the sample queues.conf (issue #8065, adamg) - -2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: sync with trunk - move variable declarations - to the beginning of a block. - -2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz> - - * main/rtp.c: Allow one-way RTP streams (device->Asterisk) - -2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org> - - * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent - build problems: - with AST_DEVMODE, building codecs/lpc10 fails - because of lots of warnings, and the configure step in editline - fails as well. Fix this by removing the -Werror in these steps. - - on FreeBSD (but probably on other platforms as well), the final - link of asterisk fails because AST_LIBS was not exported to the - subdirs Makefiles. Add a proper fix in the top-level Makefile (a - possible alternative way is to add "export AST_LIBS" near the - beginning of the file). With this fix, i believe that some of the - platform-specific conditionals in main/Makefile are redundant - (because they should be already dealt with in the top level - Makefile) but i don't have a platform to check. Merging to head - will happen in a moment. - -2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment - of previous fix: Issue #7928 - Don't send both 404 and 503. Fix - by phsultan with a small fix by me, myself or I. Thanks, - Philippe! (This was caused by my changes to the transaction - handling) - - * channels/chan_sip.c: Found some buggy SIP clients (phones Planet - VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which - sends ACK not on OK message only (when remote party answers) but - on RINGING message too, so when we send 200 OK message, we get - unidentified ACK message (because INVITE acknowledged on RINGING - message already), so 200 OK retransmits within its retransmission - interval then call gets dropped. If someone else knows how to - provide workaround for such cases, please, fix it in correct way. - Thanks to ssh from #asteriskru for provide access to his box to - study and fix this case. - -2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com> - - * agi, utils: ignore temporary files made by the Makefiles during a - build - - * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, - codecs/Makefile, utils/Makefile, configure, - build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, - Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, - pbx/Makefile, res/Makefile, channels/Makefile: fix a few build - system bugs, and convert Makefiles to be compatible with GNU make - 3.80 - -2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com> - - * main/asterisk.c, main/cli.c: Fix a bug with the removal of - 'atleast' argument to 'core verbose' and 'core debug'. Add that - argument back in. - -2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more - carefully when no CallingNumber IE available - - * channels/h323/ast_h323.cxx: Fake display name by called number on - incoming calls (until passing connected number/connected name is - not implemented) - - * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add - includes - - * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly - pass TON/PRESENTATION information - original - H323Connection::SendSignalSetup() destroys Q.931 fields. - -2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com> - - * main/Makefile: yet another place where we were not using the - correct CFLAGS by default - - * main/Makefile: missed one conversion to ASTCFLAGS - -2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass - TON/PRESENTATION information too - -2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com> - - * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, - main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, - Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse - CFLAGS and LDFLAGS for build of Asterisk components, because they - are also then used for non-Asterisk components (like menuselect); - use our own variables instead - - * configure, configure.ac: support --without-curl in configure - script - - * Makefile.rules: another cross-compile fix - - * Makefile: a couple more environment settings that can't leak into - the menuselect build - - * main/cli.c: proper fix for ast_group_t change - - * include/asterisk/lock.h: eliminate compiler warning when - DEBUG_CHANNEL_LOCKS is enabled and users of this header file - don't also include channel.h - -2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Fix incorrect argument order for member names, - on persisted members. Issue 8047, patch by jmls. - -2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c, res/res_monitor.c, - include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, - channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, - main/udptl.c, main/frame.c, funcs/func_timeout.c, - channels/chan_sip.c, apps/app_festival.c, - channels/iax2-provision.c, apps/app_alarmreceiver.c, - res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: - Put in missing \ns on the end of ast_logs (issue #7936 reported - by wojtekka) - -2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <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 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Extend call establishment timeout - -2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Make sure the pvt exists before accessing - it again as it may have gone away (issue #7562 reported by Seb7 - and issue #7939 reported by sorg) - - * main/cli.c: Warning be gone! - -2006-09-28 16:41 +0000 [r43899] BJ Weschke <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 16:17 +0000 [r43893] 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 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Fix compiler warning - -2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - - tim_ringenbach reported and patched) - - * apps/app_queue.c: Autopause not working for queue members. (#8042 - - jmls reported and patch) - -2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force - remote side to start media on outgoing PROGRESS message - - * include/asterisk/compiler.h: Put attribute tag at correct place - -2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c: fixed a bug which led to chan_list zombies, - when the call could not be properly established in misdn_call. - also removed the ACK_HDLC stuff which is not really needed. - -2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Do not open transmit channel until - TCS is received - - * main/file.c: Don't warn on HOLD/UNHOLD control frames - - * main/file.c: Don't treat unknown control frames as voice - -2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <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 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com> - - * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS - not being set under certain circumstances. Fix a minor issue, to - make it use the filenames that were parsed, instead of the entire - argument string. Fix Background() to return -1 like Playback(), - if no args are specified. - -2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Compensate for out of order packets better if RFC2833 - compensation is turned on. - - * channels/chan_iax2.c: Get rid of two functions from a time now - past (we THINK these are from pre-recursive lock time) that may - be contributing to two open issues on the bug tracker (7562/7939) - and that has the potential to just make bad things happen if the - timing is right. - -2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com> - - * main/channel.c,res/res_features.c: Fix a problem that occurred if - a user entered a digit - that matched a bridge feature that was configured using multiple - digits, and the digit that was pressed timed out in the feature - digit timeout period. For example, if blind transfer is - configured as '##', and a user presses just '#'. In this - situation, the call would lock up and no longer pass any frames. - (issue #7977 reported by festr, and issue #7982 reported by - michaels and valuable input provided by mneuhauser and kuj. Fixed - by me, with testing help and peer review from Joshua Colp). There - are a couple of issues involved in this fix: 1) When - ast_generic_bridge determines that there has been a timeout, it - returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets - this result, it calls ast_generic_bridge over again with the same - timestamp for the next event. This results in an endless loop of - nothing until the call is terminated. This is resolved by simply - changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it - sees a timeout. 2) I also changed ast_channel_bridge such that if - in the process of calculating the time until the next event, it - knows a timeout has already occured, to immediately return - AST_BRIDGE_COMPLETE instead of attempting to bridge the channels - anyway. 3) In the process of testing the previous two changes, I - ran into a problem in res_features where ast_channel_bridge would - return because it determined that there was a timeout. However, - ast_bridge_call in res_features would then determine by its own - calculation that there was still 1 ms before the timeout really - occurs. It would then proceed, and since the bridge broke out and - did *not* return a frame, it interpreted this as the call was - over and hung up the channels. The reason for this was because - ast_bridge_call in res_features and ast_channel_bridge in - channel.c were using different times for their calculations. - channel.c uses the start_time on the bridge config, which is the - time that the feature digit was recieved. However, res_features - had another time, 'start', which was set right before calling - ast_channel_bridge. 'start' will always be slightly after - start_time in the bridge config, and sometimes enough to round up - to one ms. This is fixed by making ast_bridge_call use the same - time as ast_channel_bridge for the timeout calculation. ........ - -2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn - versioning, since Asterisk has it's own - -2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make rfc2833compensate a global option. - -2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Backport revision 43754 from the trunk, - which removes an unused buffer from mm_login to close bug 8038, - as well as addresses some formatting and coding guidelines issues - in passing. Originally, I did not commit this to 1.4 since it is - not necessarily fixing a bug. However, since the IMAP storage - code is brand new, I decided it would be better to make the - change here as well, in case someone has to work on this code to - address issues in the very near future. I don't want to make - unnecessary merge problems going to the trunk. - -2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com> - - * configs/extensions.ael.sample: This change to extensions.ael was - to fix bug 8031; the install scripts are causing it to be copied - to /etc/asterisk/extensions.ael, and because it is a fairly - direct conversion of the original extensions.conf, the macro and - context names clash with the existing extensions.conf. So, I put - an ael- in front of all macros and contexts, and checked every - goto and macro call. Also, this file compiles under aelparse. - -2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Back in revision 4798, this message was changed from - using ast_cli() to directly calling write(). During this change, - checking if this was a remote console was removed. This caused - this message about using "exit" or "quit" to exit an Asterisk - console to come up in times where it did not make sense. This - change restores the check to see if this is a remote console - before printing the message. (fixes BE-65) - -2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com> - - * .cleancount, main/cli.c, channels/chan_sip.c, - include/asterisk/channel.h: Use proper type to represent the group variable - (issue #8025 reported by makoto) - -2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Add missing newline character in the warning - message about deprecated TOS values in configuration. - - * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain - mailbox definitions, don't introduce a length limit on the - definition by using a 256 byte temporary storage buffer. Instead, - make the temporary buffer just as big as it needs to be to hold - the entire mailbox definition. (fixes BE-68) - -2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: Strip options off the argument passed for - devicestate in chan_local. (issue #8034 reported by pcardozo) - - * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight - overhaul of the whisper support. 1. We need to duplicate the - frame from ast_translate 2. We need to ensure we always have - signed linear coming in for signed linear combining. 3. We need - to ensure we are always feeding signed linear out. 4. Properly - store and restore write format when beeping on the channel we are - whispering on. 5. Properly discontinue the stream on the channel - for the beep. (issue #8019 reported by timkelly1980) - -2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: update to use 1.4.3 core sounds, with corrected - beep/beeperr/tt-monkeys files - -2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com> - - * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by - Dan Austin. Maximum values were incorrect, which is why this is - being put in 1.4 - - * channels/chan_skinny.c: Add proper codec support to chan_skinny. - Works with at least ulaw, alaw, and g729a. This is technically a - "new feature", but there are justifications for it. I found a bug - with the recent rtp packetization changes, which caused the media - setup to fail under certain circumstances, particularly when - using allow=all, or having no allow= statements (globally or on - the device). I could have either removed the rtp packetization - features, or I could add proper codec support (which, without, I - think most people would consider to be a bug anyways). - -2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Should have moved these lines up in the - merge, instead of removing them - - * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) - delete=yes was ignored 2) maxmessages was ignored - -2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, - channels/h323/cisco-h225.asn: Fix ASN1 description of - non-standard Cisco extensions - - * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport - changes of trunk: 1) r43540: Avoid possible deadlock on channel - destruction 2) r43590: Disable fastStart if requested by remote - side - -2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com> - - * sounds/Makefile: One more fix for sounds installation - this time - for portability. Reported to asterisk-dev mailing list. - -2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com> - - * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from - crashing if trying to play an OGG moh file. - -2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, - channels/chan_h323.c: Merged revisions 43472,43495 from trunk - -2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com> - - * channels/iax2-provision.c: Fix a CLI command registration issue - where an erroneous message claiming that "iax2 show provisioning" - was already registered. This was because this command was - registering itself as both the command, as well as the command it - is deprecating. (issue #8022, reported by bjweeks, fixed by - myself) - - * channels/chan_iax2.c:Check to see if the channel that is activating the - IAXPEER function is actually an IAX2 channel before proceeding to - process it to avoid crashing. (issue #8017, reported by admott, - fixed by myself) - -2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't output the 'build complete' message when the - target being run is already going to do an installation - -2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded - properly. Remove reload support, since it doesn't - actually...work. - -2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This commits a change to return - MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all - goes well for bug 8004 - - * pbx/pbx_ael.c: If the extensions.ael file not found, or - unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. - -2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com> - - * main/cli.c: Make sure we explicitly set the CLI command to not be - deprecated, if it isn't. - -2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: use rebuilt extra sounds - - * main/channel.c: all the Linux systems I have don't use - '__m_count' for this field, so I don't know where this came - from... - -2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com> - - * include/asterisk/threadstorage.h: backport the compatability fix - to use attribute_malloc instaed of __attribute__ ((malloc)) - - * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN - could not be configured (issue #8006, Mithraen) - - * main/frame.c: Suppress a compiler warning about the use of a - potentially uninitialized variable. It couldn't actually happen, - though. - -2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: First shot at unload_module in - chan_skinny.. More to come. - -2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com> - - * include/asterisk/jabber.h, channels/chan_gtalk.c, - res/res_jabber.c: updates for better compontent support - -2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we - actually documented how the new features in res_odbc actually - work. (Oops) - -2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c: Some more clean up in the load function for - chan_oss (issue #8002 reported by Mithraen with minor mods by - moi) - - * channels/chan_mgcp.c: Clean up chan_mgcp's module load function - (issue #8001 reported by Mithraen with mods by moi) - -2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com> - - * main/Makefile, build_tools/strip_nonapi (added): add another - attempt to strip non-API symbols from the final binary... script - will need to be extended to work on non-Linux systems - -2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_url.c: Fix documentation to reflect how Url() really - works - - * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates - -2006-09-21 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta2 released. - -2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com> - - * main/Makefile: remove this change... it requires binutils 2.17 - -2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com> - - * build_tools/make_version: fix minor typo in the way version is - handled - -2006-09-20 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta1 released. diff --git a/apps/app_userevent.c b/apps/app_userevent.c index df7bc58a7..dd8000376 100644 --- a/apps/app_userevent.c +++ b/apps/app_userevent.c @@ -59,7 +59,7 @@ static int userevent_exec(struct ast_channel *chan, void *data) { struct ast_module_user *u; char *parse, buf[2048] = ""; - int x, buflen = 0; + int x, buflen = 0, xlen; AST_DECLARE_APP_ARGS(args, AST_APP_ARG(eventname); AST_APP_ARG(extra)[100]; @@ -77,8 +77,13 @@ static int userevent_exec(struct ast_channel *chan, void *data) AST_STANDARD_APP_ARGS(args, parse); for (x = 0; x < args.argc - 1; x++) { - ast_copy_string(buf + buflen, args.extra[x], sizeof(buf) - buflen - 2); - buflen += strlen(args.extra[x]); + /* Stop once a header comes up that exceeds our buffer. */ + if (sizeof(buf) <= buflen + (xlen = strlen(args.extra[x])) + 3) { + ast_log(LOG_WARNING, "UserEvent exceeds our buffer length! Truncating.\n"); + break; + } + ast_copy_string(buf + buflen, args.extra[x], sizeof(buf) - buflen - 3); + buflen += xlen; ast_copy_string(buf + buflen, "\r\n", 3); buflen += 2; } diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c index b0530f78c..f8dfac37c 100644 --- a/apps/app_voicemail.c +++ b/apps/app_voicemail.c @@ -5816,7 +5816,7 @@ static int vm_intro_gr(struct ast_channel *chan, struct vm_state *vms) res = ast_play_and_wait(chan, "vm-denExeteMynhmata"); return res; } - + /* Default English syntax */ static int vm_intro_en(struct ast_channel *chan, struct vm_state *vms) { @@ -5861,6 +5861,101 @@ static int vm_intro_en(struct ast_channel *chan, struct vm_state *vms) return res; } +/* Version of vm_intro() designed to work for many languages. + * + * It is hoped that this function can prevent the proliferation of + * language-specific vm_intro() functions and in time replace the language- + * specific functions which already exist. An examination of the language- + * specific functions revealed that they all corrected the same deficiencies + * in vm_intro_en() (which was the default function). Namely: + * + * 1) The vm-Old and vm-INBOX sound files were overloaded. The English + * wording of the voicemail greeting hides this problem. For example, + * vm-INBOX contains only the word "new". This means that both of these + * sequences produce valid utterances: + * * vm-youhave digit/1 vm-INBOX vm-message (you have one new message) + * * vm-press digit/1 vm-for vm-INBOX vm-messages (press 1 for new messages) + * However, if we rerecord vm-INBOX to say "the new" (which is unavoidable + * in many languages) the first utterance becomes "you have 1 the new message". + * 2) The function contains hardcoded rules for pluralizing the word "message". + * These rules are correct for English, but not for many other languages. + * 3) No attempt is made to pluralize the adjectives ("old" and "new") as + * required in many languages. + * 4) The gender of the word for "message" is not specified. This is a problem + * because in many languages the gender of the number in phrases such + * as "you have one new message" must match the gender of the word + * meaning "message". + * + * Fixing these problems for each new language has meant duplication of effort. + * This new function solves the problems in the following general ways: + * 1) Add new sound files vm-new and vm-old. These can be linked to vm-INBOX + * and vm-Old respectively for those languages where it makes sense. + * 2) Call ast_say_counted_noun() to put the proper gender and number prefix + * on vm-message. + * 3) Call ast_say_counted_adjective() to put the proper gender and number + * prefix on vm-new and vm-old (none for English). + * 4) Pass the gender of the language's word for "message" as an agument to + * this function which is can in turn pass on to the functions which + * say numbers and put endings on nounds and adjectives. + * + * All languages require these messages: + * vm-youhave "You have..." + * vm-and "and" + * vm-no "no" (in the sense of "none", as in "you have no messages") + * + * To use it for English, you will need these additional sound files: + * vm-new "new" + * vm-message "message", singular + * vm-messages "messages", plural + * + * If you use it for Russian and other slavic languages, you will need these additional sound files: + * + * vm-newn "novoye" (singular, neuter) + * vm-newx "novikh" (counting plural form, genative plural) + * vm-message "sobsheniye" (singular form) + * vm-messagex1 "sobsheniya" (first counting plural form, genative singular) + * vm-messagex2 "sobsheniy" (second counting plural form, genative plural) + * digits/1n "odno" (neuter singular for phrases such as "one message" or "thirty one messages") + * digits/2n "dva" (neuter singular) + */ +static int vm_intro_multilang(struct ast_channel *chan, struct vm_state *vms, const char message_gender[]) +{ + int res; + int lastnum = 0; + + res = ast_play_and_wait(chan, "vm-youhave"); + + if (!res && vms->newmessages) { + lastnum = vms->newmessages; + + if (!(res = ast_say_number(chan, lastnum, AST_DIGIT_ANY, chan->language, message_gender))) { + res = ast_say_counted_adjective(chan, lastnum, "vm-new", message_gender); + } + + if (!res && vms->oldmessages) { + res = ast_play_and_wait(chan, "vm-and"); + } + } + + if (!res && vms->oldmessages) { + lastnum = vms->oldmessages; + + if (!(res = ast_say_number(chan, lastnum, AST_DIGIT_ANY, chan->language, message_gender))) { + res = ast_say_counted_adjective(chan, lastnum, "vm-old", message_gender); + } + } + + if (!res) { + if (lastnum == 0) { + res = ast_play_and_wait(chan, "vm-no"); + } else { + res = ast_say_counted_noun(chan, lastnum, "vm-message"); + } + } + + return res; +} + /* Default Hebrew syntax */ static int vm_intro_he(struct ast_channel *chan, struct vm_state *vms) { @@ -6491,151 +6586,6 @@ static int vm_intro_cz(struct ast_channel *chan,struct vm_state *vms) return res; } -static int get_lastdigits(int num) -{ - num %= 100; - return (num < 20) ? num : num % 10; -} - -static int vm_intro_ru(struct ast_channel *chan,struct vm_state *vms) -{ - int res; - int lastnum = 0; - int dcnum; - - res = ast_play_and_wait(chan, "vm-youhave"); - if (!res && vms->newmessages) { - lastnum = get_lastdigits(vms->newmessages); - dcnum = vms->newmessages - lastnum; - if (dcnum) - res = say_and_wait(chan, dcnum, chan->language); - if (!res && lastnum) { - if (lastnum == 1) - res = ast_play_and_wait(chan, "digits/odno"); - else - res = say_and_wait(chan, lastnum, chan->language); - } - - if (!res) - res = ast_play_and_wait(chan, (lastnum == 1) ? "vm-novoe" : "vm-novyh"); - - if (!res && vms->oldmessages) - res = ast_play_and_wait(chan, "vm-and"); - } - - if (!res && vms->oldmessages) { - lastnum = get_lastdigits(vms->oldmessages); - dcnum = vms->oldmessages - lastnum; - if (dcnum) - res = say_and_wait(chan, dcnum, chan->language); - if (!res && lastnum) { - if (lastnum == 1) - res = ast_play_and_wait(chan, "digits/odno"); - else - res = say_and_wait(chan, lastnum, chan->language); - } - - if (!res) - res = ast_play_and_wait(chan, (lastnum == 1) ? "vm-staroe" : "vm-staryh"); - } - - if (!res && !vms->newmessages && !vms->oldmessages) { - lastnum = 0; - res = ast_play_and_wait(chan, "vm-no"); - } - - if (!res) { - switch (lastnum) { - case 1: - res = ast_play_and_wait(chan, "vm-soobshenie"); - break; - case 2: - case 3: - case 4: - res = ast_play_and_wait(chan, "vm-soobsheniya"); - break; - default: - res = ast_play_and_wait(chan, "vm-soobsheniy"); - break; - } - } - - return res; -} - -/* UKRAINIAN syntax */ -/* in ukrainian the syntax is different so we need the following files - * -------------------------------------------------------- - * /digits/ua/1e 'odne' - * vm-nove 'nove' - * vm-stare 'stare' - */ - -static int vm_intro_ua(struct ast_channel *chan,struct vm_state *vms) -{ - int res; - int lastnum = 0; - int dcnum; - - res = ast_play_and_wait(chan, "vm-youhave"); - if (!res && vms->newmessages) { - lastnum = get_lastdigits(vms->newmessages); - dcnum = vms->newmessages - lastnum; - if (dcnum) - res = say_and_wait(chan, dcnum, chan->language); - if (!res && lastnum) { - if (lastnum == 1) - res = ast_play_and_wait(chan, "digits/ua/1e"); - else - res = say_and_wait(chan, lastnum, chan->language); - } - - if (!res) - res = ast_play_and_wait(chan, (lastnum == 1) ? "vm-nove" : "vm-INBOX"); - - if (!res && vms->oldmessages) - res = ast_play_and_wait(chan, "vm-and"); - } - - if (!res && vms->oldmessages) { - lastnum = get_lastdigits(vms->oldmessages); - dcnum = vms->oldmessages - lastnum; - if (dcnum) - res = say_and_wait(chan, dcnum, chan->language); - if (!res && lastnum) { - if (lastnum == 1) - res = ast_play_and_wait(chan, "digits/ua/1e"); - else - res = say_and_wait(chan, lastnum, chan->language); - } - - if (!res) - res = ast_play_and_wait(chan, (lastnum == 1) ? "vm-stare" : "vm-Old"); - } - - if (!res && !vms->newmessages && !vms->oldmessages) { - lastnum = 0; - res = ast_play_and_wait(chan, "vm-no"); - } - - if (!res) { - switch (lastnum) { - case 1: - case 2: - case 3: - case 4: - res = ast_play_and_wait(chan, "vm-message"); - break; - default: - res = ast_play_and_wait(chan, "vm-messages"); - break; - } - } - - return res; -} - - static int vm_intro(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm_state *vms) { char prefile[256]; @@ -6675,9 +6625,9 @@ static int vm_intro(struct ast_channel *chan, struct ast_vm_user *vmu, struct vm } else if (!strcasecmp(chan->language, "no")) { /* NORWEGIAN syntax */ return vm_intro_no(chan, vms); } else if (!strcasecmp(chan->language, "ru")) { /* RUSSIAN syntax */ - return vm_intro_ru(chan, vms); + return vm_intro_multilang(chan, vms, "n"); } else if (!strcasecmp(chan->language, "ua")) { /* UKRAINIAN syntax */ - return vm_intro_ua(chan, vms); + return vm_intro_multilang(chan, vms, "n"); } else if (!strcasecmp(chan->language, "he")) { /* HEBREW syntax */ return vm_intro_he(chan, vms); } else { /* Default to ENGLISH */ diff --git a/channels/chan_local.c b/channels/chan_local.c index b2e3c72b8..9f3dd1d77 100644 --- a/channels/chan_local.c +++ b/channels/chan_local.c @@ -168,10 +168,6 @@ static int local_queue_frame(struct local_pvt *p, int isoutbound, struct ast_fra /* Recalculate outbound channel */ other = isoutbound ? p->owner : p->chan; - /* do not queue frame if generator is on both local channels */ - if (us && us->generator && other->generator) - return 0; - /* Set glare detection */ ast_set_flag(p, LOCAL_GLARE_DETECT); if (ast_test_flag(p, LOCAL_CANCEL_QUEUE)) { @@ -186,6 +182,10 @@ static int local_queue_frame(struct local_pvt *p, int isoutbound, struct ast_fra return 0; } + /* do not queue frame if generator is on both local channels */ + if (us && us->generator && other->generator) + return 0; + /* Ensure that we have both channels locked */ while (other && ast_channel_trylock(other)) { ast_mutex_unlock(&p->lock); diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 004aaaabe..cc8fafed8 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -12698,7 +12698,11 @@ static int handle_response_register(struct sip_pvt *p, int resp, char *rest, str break; case 408: /* Request timeout */ /* Got a timeout response, so reset the counter of failed responses */ - r->regattempts = 0; + if (r) { + r->regattempts = 0; + } else { + ast_log(LOG_WARNING, "Got a 408 response to our REGISTER on call %s after we had destroyed the registry object\n", p->callid); + } break; case 479: /* SER: Not able to process the URI - address is wrong in register*/ ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username,p->registry->hostname); diff --git a/include/asterisk/say.h b/include/asterisk/say.h index 17070c2c4..bc36a2a61 100644 --- a/include/asterisk/say.h +++ b/include/asterisk/say.h @@ -151,6 +151,10 @@ SAY_EXTERN int (* ast_say_datetime_from_now)(struct ast_channel *chan, time_t t, SAY_EXTERN int (* ast_say_date_with_format)(struct ast_channel *chan, time_t t, const char *ints, const char *lang, const char *format, const char *timezone) SAY_INIT(ast_say_date_with_format); +int ast_say_counted_noun(struct ast_channel *chan, int num, const char *noun); + +int ast_say_counted_adjective(struct ast_channel *chan, int num, const char *adjective, const char *gender); + #if defined(__cplusplus) || defined(c_plusplus) } #endif diff --git a/main/manager.c b/main/manager.c index c0245ad2a..7f436ad94 100644 --- a/main/manager.c +++ b/main/manager.c @@ -2142,11 +2142,15 @@ static int action_userevent(struct mansession *s, const struct message *m) { const char *event = astman_get_header(m, "UserEvent"); char body[2048] = ""; - int x, bodylen = 0; + int x, bodylen = 0, xlen; for (x = 0; x < m->hdrcount; x++) { if (strncasecmp("UserEvent:", m->headers[x], strlen("UserEvent:"))) { + if (sizeof(body) < bodylen + (xlen = strlen(m->headers[x])) + 3) { + ast_log(LOG_WARNING, "UserEvent exceeds our buffer length. Truncating.\n"); + break; + } ast_copy_string(body + bodylen, m->headers[x], sizeof(body) - bodylen - 3); - bodylen += strlen(m->headers[x]); + bodylen += xlen; ast_copy_string(body + bodylen, "\r\n", 3); bodylen += 2; } diff --git a/main/say.c b/main/say.c index 5903b4597..b6b973ccd 100644 --- a/main/say.c +++ b/main/say.c @@ -54,6 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/lock.h" #include "asterisk/localtime.h" #include "asterisk/utils.h" +#include "asterisk/app.h" /* Forward declaration */ static int wait_file(struct ast_channel *chan, const char *ints, const char *file, const char *lang); @@ -7265,6 +7266,114 @@ static int ast_say_datetime_from_now_ge(struct ast_channel *chan, time_t t, cons return res; } +/* In English, we use the plural for everything but one. For example: + * 1 degree + * 2 degrees + * 5 degrees + * The filename for the plural form is generated by appending "s". Note that + * purpose is to generate a unique filename, not to implement irregular + * declensions. Thus: + * 1 man + * 2 mans (the "mans" soundfile will of course say "men") + */ +static const char *counted_noun_ending_en(int num) +{ + if (num == 1 || num == -1) { + return ""; + } else { + return "s"; + } +} + +/* Counting of objects in slavic languages such as Russian and Ukrainian the + * rules are more complicated. There are two plural forms used in counting. + * They are the genative singular which we represent with the suffix "x1" and + * the genative plural which we represent with the suffix "x2". The base names + * of the soundfiles remain in English. For example: + * 1 degree (soudfile says "gradus") + * 2 degreex1 (soundfile says "gradusa") + * 5 degreex2 (soundfile says "gradusov") + */ +static const char *counted_noun_ending_slavic(int num) +{ + if (num < 0) { + num *= -1; + } + num %= 100; /* never pay attention to more than two digits */ + if (num >= 20) { /* for numbers 20 and above, pay attention to only last digit */ + num %= 10; + } + if (num == 1) { /* singular */ + return ""; + } + if (num > 0 && num < 5) { /* 2--5 get genative singular */ + return "x1"; + } else { /* 5--19 get genative plural */ + return "x2"; + } +} + +int ast_say_counted_noun(struct ast_channel *chan, int num, const char noun[]) +{ + char *temp; + int temp_len; + const char *ending; + if (!strcasecmp(chan->language, "ru")) { /* Russian */ + ending = counted_noun_ending_slavic(num); + } else if(!strcasecmp(chan->language, "ua")) { /* Ukrainian */ + ending = counted_noun_ending_slavic(num); + } else if(!strcasecmp(chan->language, "ua")) { /* Polish */ + ending = counted_noun_ending_slavic(num); + } else { /* English and default */ + ending = counted_noun_ending_en(num); + } + temp = alloca((temp_len = (strlen(noun) + strlen(ending) + 1))); + snprintf(temp, temp_len, "%s%s", noun, ending); + return ast_play_and_wait(chan, temp); +} + +/* + * In slavic languages such as Russian and Ukrainian the rules for declining + * adjectives are simpler than those for nouns. When counting we use only + * the singular (to which we give no suffix) and the genative plural (which + * we represent by adding an "x"). Oh, an in the singular gender matters + * so we append the supplied gender suffix ("m", "f", "n"). + */ +static const char *counted_adjective_ending_ru(int num, const char gender[]) +{ + if (num < 0) { + num *= -1; + } + num %= 100; /* never pay attention to more than two digits */ + if (num >= 20) { /* at 20 and beyond only the last digit matters */ + num %= 10; + } + if (num == 1) { + return gender ? gender : ""; + } else { /* all other numbers get the genative plural */ + return "x"; + } +} + +int ast_say_counted_adjective(struct ast_channel *chan, int num, const char adjective[], const char gender[]) +{ + char *temp; + int temp_len; + const char *ending; + if (!strcasecmp(chan->language, "ru")) { /* Russian */ + ending = counted_adjective_ending_ru(num, gender); + } else if (!strcasecmp(chan->language, "ua")) { /* Ukrainian */ + ending = counted_adjective_ending_ru(num, gender); + } else if (!strcasecmp(chan->language, "pl")) { /* Polish */ + ending = counted_adjective_ending_ru(num, gender); + } else { /* English and default */ + ending = ""; + } + temp = alloca((temp_len = (strlen(adjective) + strlen(ending) + 1))); + snprintf(temp, temp_len, "%s%s", adjective, ending); + return ast_play_and_wait(chan, temp); +} + /* diff --git a/res/res_features.c b/res/res_features.c index aaa1ff593..c7fa2ef10 100644 --- a/res/res_features.c +++ b/res/res_features.c @@ -302,7 +302,7 @@ static int metermaidstate(const char *data) return AST_DEVICE_INUSE; } -static int park_call_full(struct ast_channel *chan, struct ast_channel *peer, int timeout, int *extout, char *orig_chan_name) +static int park_call_full(struct ast_channel *chan, struct ast_channel *peer, int timeout, int *extout, const char *orig_chan_name) { struct parkeduser *pu, *cur; int i, x = -1, parking_range, parkingnum_copy; @@ -393,7 +393,7 @@ static int park_call_full(struct ast_channel *chan, struct ast_channel *peer, in struct ast_channel *tmpchan, *base_peer; char other_side[AST_CHANNEL_NAME]; char *c; - ast_copy_string(other_side, peer->name, sizeof(other_side)); + ast_copy_string(other_side, S_OR(orig_chan_name, peer->name), sizeof(other_side)); if ((c = strrchr(other_side, ','))) { *++c = '1'; } @@ -404,7 +404,7 @@ static int park_call_full(struct ast_channel *chan, struct ast_channel *peer, in ast_channel_unlock(tmpchan); } } else { - ast_copy_string(pu->peername, peer->name, sizeof(pu->peername)); + ast_copy_string(pu->peername, S_OR(orig_chan_name, peer->name), sizeof(pu->peername)); } } @@ -484,11 +484,10 @@ int ast_park_call(struct ast_channel *chan, struct ast_channel *peer, int timeou return park_call_full(chan, peer, timeout, extout, NULL); } -static int masq_park_call(struct ast_channel *rchan, struct ast_channel *peer, int timeout, int *extout, int play_announcement) +static int masq_park_call(struct ast_channel *rchan, struct ast_channel *peer, int timeout, int *extout, int play_announcement, const char *orig_chan_name) { struct ast_channel *chan; struct ast_frame *f; - char *orig_chan_name = NULL; int park_status; /* Make a new, fake channel that we'll use to masquerade in the real one */ @@ -514,7 +513,7 @@ static int masq_park_call(struct ast_channel *rchan, struct ast_channel *peer, i peer = chan; } - if (!play_announcement) { + if (!play_announcement || !orig_chan_name) { orig_chan_name = ast_strdupa(chan->name); } @@ -530,12 +529,12 @@ static int masq_park_call(struct ast_channel *rchan, struct ast_channel *peer, i int ast_masq_park_call(struct ast_channel *rchan, struct ast_channel *peer, int timeout, int *extout) { - return masq_park_call(rchan, peer, timeout, extout, 0); + return masq_park_call(rchan, peer, timeout, extout, 0, NULL); } -static int masq_park_call_announce(struct ast_channel *rchan, struct ast_channel *peer, int timeout, int *extout) +static int masq_park_call_announce(struct ast_channel *rchan, struct ast_channel *peer, int timeout, int *extout, const char *orig_chan_name) { - return masq_park_call(rchan, peer, timeout, extout, 1); + return masq_park_call(rchan, peer, timeout, extout, 1, orig_chan_name); } #define FEATURE_RETURN_HANGUP -1 #define FEATURE_RETURN_SUCCESSBREAK 0 @@ -569,10 +568,12 @@ static int builtin_parkcall(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *parkee; int res = 0; struct ast_module_user *u; + const char *orig_chan_name; u = ast_module_user_add(chan); set_peers(&parker, &parkee, peer, chan, sense); + orig_chan_name = ast_strdupa(parker->name); /* we used to set chan's exten and priority to "s" and 1 here, but this generates (in some cases) an invalid extension, and if "s" exists, could errantly @@ -587,7 +588,7 @@ static int builtin_parkcall(struct ast_channel *chan, struct ast_channel *peer, res = ast_safe_sleep(chan, 1000); if (!res) { /* one direction used to call park_call.... */ - masq_park_call_announce(parkee, parker, 0, NULL); + masq_park_call_announce(parkee, parker, 0, NULL, orig_chan_name); res = 0; /* PBX should hangup zombie channel */ } @@ -742,7 +743,7 @@ static int builtin_blindtransfer(struct ast_channel *chan, struct ast_channel *p res = finishup(transferee); if (res) res = -1; - else if (!masq_park_call_announce(transferee, transferer, 0, NULL)) { /* success */ + else if (!masq_park_call_announce(transferee, transferer, 0, NULL, NULL)) { /* success */ /* We return non-zero, but tell the PBX not to hang the channel when the thread dies -- We have to be careful now though. We are responsible for hanging up the channel, else it will never be hung up! */ @@ -2065,6 +2066,7 @@ static int park_call_exec(struct ast_channel *chan, void *data) /* Cache the original channel name in case we get masqueraded in the middle * of a park--it is still theoretically possible for a transfer to happen before * we get here, but it is _really_ unlikely */ + char *orig_chan_name = ast_strdupa(chan->name); char orig_exten[AST_MAX_EXTENSION]; int orig_priority = chan->priority; @@ -2089,7 +2091,7 @@ static int park_call_exec(struct ast_channel *chan, void *data) res = ast_safe_sleep(chan, 1000); /* Park the call */ if (!res) { - res = masq_park_call_announce(chan, chan, 0, NULL); + res = masq_park_call_announce(chan, chan, 0, NULL, orig_chan_name); /* Continue on in the dialplan */ if (res == 1) { ast_copy_string(chan->exten, orig_exten, sizeof(chan->exten)); |