diff options
| author | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-12-02 19:34:13 +0000 |
|---|---|---|
| committer | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-12-02 19:34:13 +0000 |
| commit | e4e6e152c5c6d7f1c85c8ed506d90ef838ad03af (patch) | |
| tree | 1fca640326ed77efd78f9b5696d06dca7f75af55 /ChangeLog | |
| parent | d0d8544a57560fc2096b50416aa8128b50301e29 (diff) | |
| parent | 665404ff2a9ba2d3e8585b0c622c97690cb6925f (diff) | |
Create Asterisk 1.6.2.15 from 1.6.2.15-rc1
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.15@297393 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 909 |
1 files changed, 909 insertions, 0 deletions
@@ -1,3 +1,912 @@ +2010-11-15 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.15-rc1 + +2010-11-15 18:24 +0000 [r294988-295062] Tilghman Lesher <tlesher@digium.com> + + * tests/test_expr.c (added), /: Merged revisions 295026 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) + | 2 lines Create test verifying results of expression parser + ........ + + * funcs/func_curl.c: It is possible to crash Asterisk by feeding + the curl engine invalid data. (closes issue #18161) Reported by: + wdoekes Patches: 20101029__issue18161.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman + +2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c: Return correct error code if lock path + fails. The recent changes to open_mailbox actually caused it to + be fixed, but let's be consistent. Reported by alecdavis in + asterisk-dev. + + * apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 + Nov 2010) | 16 lines Fix regression causing abort in voicemail + after opening a mailbox with no mesgs. In order to be more safe, + some error handling code was changed to respect more error + conditions including the potential memory allocation failure for + deleted and heard message tracking introduced in 293004. However, + last_message_index returns -1 for zero messages (perhaps as + expected) and was triggering the stricter error checking. Because + last_message_index is only called directly in one place, just + return 0 from open_mailbox (for file based storage) when no + messages are detected unless a real error has occurred. (closes + issue #18240) Reported by: leobrown Patches: + bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) + Tested by: pabelanger ........ + +2010-11-12 02:44 +0000 [r294822] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 + Nov 2010) | 11 lines Asterisk is getting a "No D-channels + available!" warning message every 4 seconds. Asterisk is just + whining too much with this message: "No D-channels available! + Using Primary channel XXX as D-channel anyway!". Filtered the + message so it only comes out once if there is no D channel + available without an intervening D channel available period. + (closes issue #17270) Reported by: jmls ........ + +2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) + | 18 lines Fix problem with qualify option packets for realtime + peers never stopping. The option packets not only never stopped, + but if a realtime peer was not in the peer list multiple options + dialogs could accumulate over time. This scenario has the + potential to progress to the point of saturating a link just from + options packets. The fix was to ensure that the poke scheduler + checks to see if a peer is in the peer list before continuing to + poke. The reason a peer must be in the peer list to be able to + properly manage an options dialog is because otherwise the call + pointer is lost when the peer is regenerated from the database, + which is how existing qualify dialogs are detected. (closes issue + #16382) (closes issue #17779) Reported by: lftsy Patches: + bug16382-3.patch uploaded by jpeeler (license 325) Tested by: + zerohalo ........ + + * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged + revisions 294384 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) + | 47 lines Fix a deadlock in device state change processing. + Copied from some notes from the original author (Russell): + Deadlock scenario: Thread 1: device state change thread Holds - + rdlock on contexts Holds - hints lock Waiting on channels + container lock Thread 2: SIP monitor thread Holds the "iflock" + Holds a sip_pvt lock Holds channel container lock Waiting for a + channel lock Thread 3: A channel thread (chan_local in this case) + Holds 2 channel locks acquired within app_dial Holds a 3rd + channel lock it got inside of chan_local Holds a local_pvt lock + Waiting on a rdlock of the contexts lock A bunch of other threads + waiting on a wrlock of the contexts lock To address this + deadlock, some locking order rules must be put in place and + enforced. Existing relevant rules: 1) channel lock before a pvt + lock 2) contexts lock before hints lock 3) channels container + before a channel What's missing is some enforcement of the order + when you involve more than any two. To fix this problem, I put in + some code that ensures that (at least in the code paths involved + in this bug) the locks in (3) come before the locks in (2). To + change the operation of thread 1 to comply, I converted the + storage of hints to an astobj2 container. This allows processing + of hints without holding the hints container lock. So, in the + code path that led to thread 1's state, it no longer holds either + the contexts or hints lock while it attempts to lock the channels + container. (closes issue #18165) Reported by: antonio ABE-2583 + ........ + +2010-11-10 23:16 +0000 [r294571] Tilghman Lesher <tlesher@digium.com> + + * main/features.c: Actually pay attention to documented settings in + features.conf. (closes issue #16757) Reported by: voxter Patches: + 20101012__issue16757.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/994/ + +2010-11-10 12:41 +0000 [r294500] Russell Bryant <russell@digium.com> + + * main/devicestate.c: Improve a debug message to be more readable + and consistent. (closes issue #18282) Reported by: klaus3000 + Patches: ast_devstate2str-patch.txt uploaded by klaus3000 + (license 65) + +2010-11-09 20:27 +0000 [r294429] Tilghman Lesher <tlesher@digium.com> + + * configure, configure.ac: Detect GMime properly on systems where + gmime flags and libs are configured with pkg-config. (closes + issue #16155) Reported by: jcollie Patches: + 20100917__issue16155.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman + +2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler <jpeeler@digium.com> + + * res/res_timing_timerfd.c: add missing unlock not present in + 294277 + + * main/timing.c, main/channel.c, res/res_timing_timerfd.c, + include/asterisk/timing.h: Fix playback failure when using IAX + with the timerfd module. To fix this issue the alert pipe will + now be used when the timerfd module is in use. There appeared to + be a race that was not solved by adding locking in the timerfd + module, but needed to be there anyway. The race was between the + timer being put in non-continuous mode in ast_read on the channel + thread and the IAX frame scheduler queuing a frame which would + enable continuous mode before the non-continuous mode event was + read. This race for now is simply avoided. (closes issue #18110) + Reported by: tpanton Tested by: tpanton I put tested by tpanton + because it was tested on his hardware. Thanks for the remote + access to debug this issue! + +2010-11-08 20:50 +0000 [r294242] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_sip.c: Go off hold when we get an empty reinvite + telling us to. (closes issue 0014448) Reported by: frawd (closes + issue #17878) Reported by: frawd + +2010-11-05 00:06 +0000 [r293969] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 + Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio + when receiving unexpected frame sizes. dahdi-linux 2.4.0 + (specifically commit 9034) added the capability for the wctc4xxp + to return more than a single packet of data in response to a + read. However, when decoding packets, codec_dahdi was still + assuming that the default number of samples was in each read. In + other words, each packet your provider sent you, regardless of + size, would result in 20 ms of decoded data (30 ms if decoding + G723). If your provider was sending 60 ms packets then + codec_dahdi would end up stripping 40 ms of data from each + transcoded frame resulting in "choppy" audio. This would only + affect systems where G729 packets are arriving in sizes greater + than 20ms or G723 packets arriving in sizes greater than 30ms. + DAHDI-744. ........ + +2010-11-03 18:31 +0000 [r293806] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 + Nov 2010) | 20 lines Party A in an analog 3-way call would + continue to hear ringback after party C answers. All parties are + analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A + flash hooks to bring C into 3-way call before C answers. (A and B + hear ringback) 4) C answers 5) A continues to hear ringback + during the 3-way call. (All parties can hear each other.) * Fixed + use of wrong variable in dahdi_bridge() that stopped ringback on + the wrong subchannel. * Made several debug messages have more + information. A similar issue happens if B and C are SIP channels. + B continues to hear ringback. For some reason this only affects + v1.8 and trunk. * Don't start ringback on the real and 3-way + subchannels when creating the 3-way conference. Removing this + code is benign on v1.6.2 and earlier. ........ + +2010-11-02 23:07 +0000 [r293723] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) + | 8 lines Add enabled/disabled information for rtautoclear sip + show settings output. When setting to zero/"no", the numeric + default was shown making it not obvious the disabled setting was + respected. (closes issue #18123) Reported by: zerohalo ........ + +2010-11-02 21:26 +0000 [r293647] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 + Nov 2010) | 6 lines Make warning message have more useful + information in it. Change "Unable to get index, and nullok is not + asserted" to "Unable to get index for '<channel-name>' on channel + <number> (<function>(), line <number>)". ........ + +2010-10-30 01:49 +0000 [r293340-293417] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 + Oct 2010) | 1 line Remove some more code that serves no purpose. + ........ + + * channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 + Oct 2010) | 1 line Remove some code that serves no purpose. + ........ + +2010-10-28 19:54 +0000 [r293195-293196] Tilghman Lesher <tlesher@digium.com> + + * main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) + | 5 lines "!00" evaluated as false, which is incorrect. Fixing. + Reported (though the reporter did not understand he was reporting + a bug) on the asterisk-users list: + http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html + ........ + + * /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c, + res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) + | 5 lines "!00" evaluated as false, which is incorrect. Fixing. + Reported (though the reporter did not understand he was reporting + a bug) on the asterisk-users list: + http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html + ........ + +2010-10-28 16:09 +0000 [r293158] Jeff Peeler <jpeeler@digium.com> + + * funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically + when you're using characters above \x7f or invalid character + escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes + Patches: issue18060_func_strings_filter_infinite_loop.patch + uploaded by wdoekes (license 717) Tested by: wdoekes + +2010-10-26 18:33 +0000 [r293118] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 + Oct 2010) | 29 lines Fix inprocess_container in voicemail to + correctly restrict max messages. The comparison function logic + was off, so the number of sessions for a given mailbox were not + being incremented properly. This problem caused the maximum + number of messages per folder to not be respected when + simultaneously leaving multiple voicemails just below the + threshold. These problems should be fixed by the above, but just + in case: Fixed resequence_mailbox to rely on the actual number of + detected number of files in a directory rather than just assuming + only 10 messages more than the maximum had been left. Also if + more messages than the maximum are deleted they are actually + removed now. The second purpose of this commit should have been + separated out probably, but is related to the above. Again, if + the number of messages in a given voicemail folder exceeds the + maximum set limit make sure to allocate enough space for the + deleted and heard index tracking array. A few random fixes: There + was a forgotten decrement of the inprocess count in + imap_store_file. When using IMAP storage, do not look in the + directory where file based storage messages may still reside and + influence the message count. Ensure to use only the first format + in sendmail. ABE-2516 ........ + +2010-10-25 19:06 +0000 [r292867] David Vossel <dvossel@digium.com> + + * channels/chan_local.c, /: Merged revisions 292866 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 + Oct 2010) | 27 lines This patch turns chan_local pvts into + astobj2 objects. chan_local does some dangerous things involving + deadlock avoidance. tech_pvt functions like hangup and + queue_frame are provided with a locked channel upon entry. Those + functions are completely safe as long as you don't attempt to + give up that channel lock, but that is impossible to guarantee + due to the required deadlock avoidance necessary to lock both the + tech_pvt and both channels involved. In the past, we have tried + to account for this by doing things like setting a "glare" flag + that indicates what function should destroy the pvt. This was + used in local_hangup and local_queue_frame to decided who should + destroy the pvt if they collided in separate threads. I have + removed the need to do this by converting all chan_local + tech_pvts to astobj2. This means we can ref a pvt before deadlock + avoidance and not have to worry about that pvt possibly getting + destroyed under us. It also cleans up where we destroy the + tech_pvt. The only unlink from the tech_pvt container occurs in + local_hangup now, which is where it should occur. Since there + still may be thread collisions on some functions like + local_hangup after deadlock avoidance, I have added some checks + to detect those collisions and exit appropriately. I think this + patch is going to solve quite a bit of weirdness we have had with + local channels in the past. ........ + +2010-10-22 21:16 +0000 [r292786] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/asterisk.ldif, channels/chan_sip.c, + configs/res_ldap.conf.sample: Update the LDIF file for LDAP. The + LDIF file asterisk.ldif was quite a bit out of date from the + asterisk.ldap-schema file, so I've now updated that to be in + sync. The asterisk.ldif file being out of sync was a problem on + my systems where I was doing an ldapadd to import the schema into + the LDAP database, and the existing file would cause problems and + ERROR messages when registering. Additional documention has been + added based on feedback in the issue I'm closing. (closes issue + #13861) Reported by: scramatte Patches: ldap-update.txt uploaded + by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, + rgenthner + +2010-10-21 13:11 +0000 [r292556] Leif Madsen <lmadsen@digium.com> + + * configs/res_ldap.conf.sample: Change res_ldap.sample.conf to + match the schema. (closes issue #17376) Reported by: jcovert + Patches: res_ldap.conf.sample.patch uploaded by jcovert (license + 551) + +2010-10-21 00:05 +0000 [r292412] Paul Belanger <paul.belanger@polybeacon.com> + + * apps/app_dial.c, /: Merged revisions 292411 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct + 2010) | 10 lines Record priv-recordintro as sln, not gsm This + removes the gsm->sln step when transcoding priv-recordintro. + (closes issue #18176) Reported by: pabelanger Patches: + chan_sip.diff uploaded by pabelanger (license 224) ........ + +2010-10-18 22:01 +0000 [r292229] Leif Madsen <lmadsen@digium.com> + + * sounds/Makefile: Fix typo in the sounds/Makefile. (Issue #17426) + +2010-10-18 21:54 +0000 [r292226] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 292223 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 + Oct 2010) | 11 lines Fix improper operator key acceptance and + clean up temp recording files. This is a fix for when pressing + the operator key after recording an unavailable, busy, name, or + temporary message in mailbox options. The operator key should not + be accepted here, but should be allowed during the message + recording. If the operator key is pressed during ensure the file + is saved or deleted as apporopriate. Also, ensure removal of + temporary recorded files after an early hang up or when message + acceptance confirmation times out. ABE-2518 ........ + +2010-10-18 21:50 +0000 [r292224] Leif Madsen <lmadsen@digium.com> + + * sounds/Makefile, /, sounds/sounds.xml: Merged revisions 292222 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) + | 9 lines Add support for the new English (Australian Accent) + sound files. (closes issue #17426) Reported by: camsown Patches: + core-sounds-en_AU.txt uploaded by camsown (license 1050) + add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested + by: camsown, lmadsen, jtodd, qwell ........ + +2010-10-16 10:03 +0000 [r292049] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * res/res_musiconhold.c, configs/musiconhold.conf.sample: Base + directory for MOH should be ASTDATADIR If the directive + 'directory' is relative, make it relative to the datadir, rather + than to the varlibdir. In the sample configuration it is relative + ('moh'). This has no effect unless you have actively set the + datadir explicitly (at build time or at run time). (closes issue + #16906) Patches: moh_datadir uploaded by tzafrir (license 46) + Review: https://reviewboard.asterisk.org/r/974/ + +2010-10-15 19:35 +0000 [r291939] Paul Belanger <paul.belanger@polybeacon.com> + + * configs/gtalk.conf.sample, /: Merged revisions 291938 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct + 2010) | 2 lines Clean up formatting. ........ + +2010-10-15 16:16 +0000 [r291904] Terry Wilson <twilson@digium.com> + + * res/res_jabber.c: Don't crash or deadlock on module unload We + can't hold the lock while pthread_join is called since + aji_log_hook will attempt to lock from the other therad. We + reorder the pthread_join and ast_aji_disconnect so that we don't + do an SSL_read() while SSL_shutdown is running, causing a crash. + +2010-10-13 23:36 +0000 [r291655] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 291643 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 + Oct 2010) | 20 lines Deadlock between dahdi_exception() and + dahdi_indicate(). There is a deadlock between dahdi_exception() + and dahdi_indicate() for analog ports. The call-waiting and + three-way-calling feature can experience deadlock if these + features are trying to do something and an event from the bridged + channel happens at the same time. Deadlock avoidance code added + to obtain necessary channel locks before attemting an operation + with call-waiting and three-way-calling. (closes issue #16847) + Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch + uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch + uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch + uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett + Review: https://reviewboard.asterisk.org/r/971/ ........ + +2010-10-13 22:58 +0000 [r291580] Terry Wilson <twilson@digium.com> + + * main/channel.c, /: Merged revisions 291577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) + | 21 lines Don't ignore frames that have been queued when + softhangup'd When an outgoing call is answered and hung up by the + far end *very* quickly, we may not read any frames and therefor + end up with a call that displays the wrong + disposition/DIALSTATUS. The reason is because ast_queue_hangup() + immediately sets the _softhangup flag on the channel and then + queues the HANGUP control frame, but __ast_read refuses to read + any frames if ast_check_hangup() indicates that a hangup request + has been made (which it will if _softhangup is set). So, we end + up losing control frames. This change makes __ast_read continue + to read frames even if a soft hangup has been requested. It + queues a hangup frame to make sure that __ast_read() will still + eventually return NULL. Much thanks to David Vossel for all of + the reviews, discussion, and help! (closes issue #16946) Reported + by: davidw Review: https://reviewboard.asterisk.org/r/740/ + ........ + +2010-10-13 15:29 +0000 [r291393] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 291392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) + | 6 lines Lock pvt so pvt->owner can't disappear when queueing up + a frame. This fixes a crash due to a hangup race condition. + ABE-2601 ........ + +2010-10-12 17:20 +0000 [r291280] Leif Madsen <lmadsen@digium.com> + + * configs/phoneprov.conf.sample: Add undocumented variables to + phoneprov.conf.sample (closes issue #18107) Reported by: lathama + Patches: phoneprov.conf.sample.diff uploaded by lathama (license + 1028) + +2010-10-12 17:05 +0000 [r291264] Tilghman Lesher <tlesher@digium.com> + + * /, main/acl.c: Merged revisions 291263 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) + | 2 lines Oops, incorrect range (although unallocated at ARIN) + ........ + +2010-10-12 16:07 +0000 [r291229] Leif Madsen <lmadsen@digium.com> + + * configs/manager.conf.sample: Add documention that mentions + options are defined but not used. (Issue #18101) + +2010-10-11 18:39 +0000 [r291073-291111] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_sip.c: Make exit from handle_request_do() + consistent. + + * /, channels/chan_sip.c: Merged revisions 291109 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) + | 1 line Add missing unlock to an exception condition in + reload_config(). ........ + + * main/cli.c: Fixed infinite loop in verbose/debug message output. + Setting the module/filename specific message level and then + changing it resulted in the linked list being looped on itself. + Traversing this linked list is an infinite loop if what you are + looking for is not in the list. Also plugged some CLI parsing + holes in the associated CLI command: * Removing a nonexistent + module from the list actually added it with a level of zero. * + Setting the non-module specific level to zero is now equivalent + to setting it to "off" as documented. + +2010-10-08 02:45 +0000 [r290863] Jeff Peeler <jpeeler@digium.com> + + * main/asterisk.c, /: Merged revisions 290862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) + | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed + at control console. A recent change was made to avoid a race + condition on shutdown which only called the end functions from + the console thread. However, when pressing Ctrl-C the quit + handler is called from the signal handler thread. (closes issue + #17698) Reported by: jmls ........ + +2010-10-07 20:57 +0000 [r290751] Jason Parker <jparker@digium.com> + + * autoconf/ast_ext_lib.m4, /, configure, + include/asterisk/autoconfig.h.in: Merged revisions 290750 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | + 9 lines Allow PRI to build properly when using --with-pri. Use + the directories found for the parent when using lib dependencies. + (closes issue #17314) Reported by: tzafrir Patches: + 17314-withdeps.diff uploaded by qwell (license 4) ........ + +2010-10-07 10:53 +0000 [r290712] Russell Bryant <russell@digium.com> + + * main/pbx.c: Don't crash when Set() is called without a value. + Review: https://reviewboard.asterisk.org/r/949/ + +2010-10-06 13:48 +0000 [r290396-290575] Tilghman Lesher <tlesher@digium.com> + + * main/file.c: Allow streaming audio from a pipe. (closes issue + #18001) Reported by: jamicque Patches: + 20100926__issue18001.diff.txt uploaded by tilghman (license 14) + Tested by: jamicque + + * res/res_jabber.c, /: Merged revisions 290392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) + | 8 lines Fix a crash by ensuring that we don't alter memory + after it's freed. (closes issue #17387) Reported by: jmls + Patches: 20100726__issue17387.diff.txt uploaded by tilghman + (license 14) Tested by: jmls ........ + +2010-10-05 19:54 +0000 [r290375] David Vossel <dvossel@digium.com> + + * apps/app_directed_pickup.c: Fixes PickupChan() not working with + full channel name. (closes issue #18011) Reported by: schern + Patches: app_directed_pickup.c.2.patch uploaded by schern + (license 995) app_directed_pickup.c.trunk.patch uploaded by + schern (license 995) Tested by: schern, dvossel + +2010-10-05 17:42 +0000 [r290324] Richard Mudgett <rmudgett@digium.com> + + * contrib/valgrind.supp, /: Merged revisions 290323 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r290323 | rmudgett | 2010-10-05 12:41:18 -0500 + (Tue, 05 Oct 2010) | 11 lines Merged revision 258974 from + https://origsvn.digium.com/svn/asterisk/trunk .......... r258974 + | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4 + lines Line 24 missed in compatibility fix in revision 233577 + added a "fun:" prefix line 24 .......... ................ + +2010-10-04 23:14 +0000 [r290101-290254] Tilghman Lesher <tlesher@digium.com> + + * pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c, + pbx/ael/ael-test/ref.ael-vtest17, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5: + Change new pattern matcher to regard dashes the same as the old + pattern matcher -- as visual candy to be ignored. Also change the + AEL parser to not generate dashes within extensions, as those + dashes would be ignored. Update the AEL tests to match this + behavior. (closes issue #17366) Reported by: murf Patches: + 20100727__issue17366.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman + + * /, configure, configure.ac: Merged revisions 290177 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 + Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........ + + * /, configure, configure.ac: Merged revisions 290100 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 + Oct 2010) | 2 lines Automatically re-run configure test for + menuselect, when the relevant makeopts settings change. ........ + +2010-10-02 08:52 +0000 [r289950] Olle Johansson <oej@edvina.net> + + * main/manager.c, /: Merged revisions 289949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 + lines Add documentation for undocumented option to AMI action + originate ........ + +2010-10-02 04:45 +0000 [r289874] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 289873 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 + Oct 2010) | 8 lines When forwarding a message, a prepend means + that the filesystem will always have a better copy. (closes issue + #17803) Reported by: dpetersen Patches: + 20100923__issue17803.diff.txt uploaded by tilghman (license 14) + Tested by: dpetersen ........ + +2010-10-01 23:01 +0000 [r289798] Jeff Peeler <jpeeler@digium.com> + + * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: + Merged revisions 289797 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) + | 15 lines Change RFC2833 DTMF event duration on end to report + actual elapsed time. The scenario here is with a non P2P early + media session. The reported time length of DTMF presses are + coming up short when sending to the remote side. Currently the + event duration is a running total that is incremented when + sending continuation packets. These continuation packets are only + triggered upon incoming media from the remote side, which means + that the running total probably is not going to end up matching + the actual length of time Asterisk received DTMF. This patch + changes the end event duration to be lengthened if it is detected + that the end event is going to come up short. Review: + https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ + +2010-10-01 17:09 +0000 [r289704] Paul Belanger <paul.belanger@polybeacon.com> + + * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions + 289703 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct + 2010) | 6 lines Disable debugging by default and reformat .config + file. Review: https://reviewboard.asterisk.org/r/929/ ........ + +2010-10-01 16:21 +0000 [r289700] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 289699 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) + | 14 lines Ensure user portion of SIP URI matches dialplan when + using encoded characters. This commit takes a simliar approach to + 288112 and checks the dialplan to determine the proper action for + an incoming contact header as to whether or not it should be + decoded or not. sip_new was blindly always decoding the + extension, which also caused the outgoing contact header to be + incorrect as well as failing to match the encoded extension in + the dialplan. (closes issue #17892) Reported by: wdoekes Patches: + bug17892-1.patch uploaded by jpeeler (license 325) Tested by: + wdoekes ........ + +2010-10-01 09:42 +0000 [r289622] schmitds <schmitds@localhost>: + + * channels/chan_sip.c: don't iterate through all dialogs to find + and delete old subscribes On every incoming subscribe there is a + iteration through all dialogs to find old subscribes and delete + them. This is slow and not RFC conform. This was only needed in + 1.2 cause a subscribe was not deleted when a dialog was + destroyed, after 1.4 a subscribe get removed when its dialog is + destroyed. (closes issue #17950) Reported by: schmidts Tested by: + schmidts Review: https://reviewboard.asterisk.org/r/901/ + +2010-09-30 19:51 +0000 [r289553] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_sip.c: Properly handle channel allocation failures + duing invites with replaces. ABE-2588 + +2010-09-30 17:09 +0000 [r289501] Brett Bryant <bbryant@digium.com> + + * /, res/res_agi.c: Merged revisions 289500 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010) + | 11 lines res_agi.c:handle_getvariablefull() could recursively + lock a channel and not release it if an argument is the current + channel's name. (closes issue #17970) Reported by: mdu113 + Patches: res_agi.c.diff3 uploaded by mdu113 (license 582) Tested + by: mdu113 Review: https://reviewboard.asterisk.org/r/947/ + ........ + +2010-09-30 15:37 +0000 [r289425] Russell Bryant <russell@digium.com> + + * /, apps/app_sms.c: Merged revisions 289424 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) + | 8 lines Fix a crash in app_sms. Since the data being passed to + the generator callback is on the stack of the SMS() application, + we must ensure that the generator is stopped before the + application exits. ABE-2587 ........ + +2010-09-29 21:03 +0000 [r289339] Jason Parker <jparker@digium.com> + + * main/channel.c, /, main/features.c: Merged revisions 289338 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | + 8 lines Allow a manager originate to succeed on forwarded + devices. The timeout to wait for an answer was being set to 0 + when a device forwarded to another extension. We don't always + need the timeout set like this, so make it an optional parameter, + and don't use it in this case. ABE-2544 ........ + +2010-09-29 20:24 +0000 [r289334] Leif Madsen <lmadsen@digium.com> + + * configs/res_ldap.conf.sample: Update sample documentation to note + md5secret requirements. + +2010-09-29 20:15 +0000 [r289332] Russell Bryant <russell@digium.com> + + * res/res_config_ldap.c: Don't completely ignore md5secret from + LDAP if the value does not begin with {md5}. This fixes a problem + that lmadsen ran in to where md5secret was not working for him. + +2010-09-29 15:04 +0000 [r289178] Matthew Nicholson <mnicholson@digium.com> + + * main/channel.c, /: Merged revisions 289177 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep + 2010) | 8 lines Set the caller id on CDRs when it is set on the + parent channel. (closes issue #17569) Reported by: tbelder + Patches: 17569.diff uploaded by tbelder (license 618) Tested by: + tbelder ........ + +2010-09-28 18:14 +0000 [r289095] Brett Bryant <bbryant@digium.com> + + * main/channel.c, /: Merged revisions 289094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) + | 14 lines Fixes an issue with the Newchannel AMI event during + the Masquerading process. Fixes an issue with the Newchannel AMI + event during the Masquerading process, where no Newchannel AMI + event was generated for the psuedo channel used during the + masquerading process. (closes issue #17987) Reported by: + RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish + (license 1122) Tested by: RadicAlish Review: + https://reviewboard.asterisk.org/r/937/ ........ + +2010-09-24 15:37 +0000 [r288747] Terry Wilson <twilson@digium.com> + + * channels/chan_local.c, /: Merged revisions 288746 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 + Sep 2010) | 5 lines Don't fail a masquerade if it is already + being hung up This avoids noise on some Local channel situations + where we don't use /n. Thanks to Alec Davis for the suggestion. + ........ + +2010-09-24 13:53 +0000 [r288637-288712] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_strings.c: Solaris won't printf a NULL. (closes issue + #18041) Reported by: asgaroth + + * cdr/cdr_pgsql.c, /, configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, main/strcompat.c, configure.ac, + include/asterisk/channel.h: Merged revisions 288636 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 + Sep 2010) | 2 lines Solaris compatibility fixes ........ + +2010-09-22 23:10 +0000 [r288500] Terry Wilson <twilson@digium.com> + + * channels/chan_local.c, /: Merged revisions 288499 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 + Sep 2010) | 8 lines Don't let a Local channel get bridged to + itself If a local channel gets bridged to itself, it becomes + orphaned with no devices left to actually tell it to hang up. + This patch modifies local_fixup() to detect this case and deny + it. Review: https://reviewboard.asterisk.org/r/934 ........ + +2010-09-22 17:49 +0000 [r288344-288417] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 288416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) + | 5 lines RFC3261 section 12.2 explicitly says out of order + requests are responded with a 500 Server Internal Error response. + ABE-2458 ........ + + * /, channels/chan_sip.c: Merged revisions 288343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) + | 2 lines During check_pendings, if the dialog is terminated with + a CANCEL, change the invitestate to INV_CANCEL like in + sip_hangup. ........ + +2010-09-22 16:44 +0000 [r288340] Russell Bryant <russell@digium.com> + + * main/asterisk.c, /: Merged revisions 288339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) + | 11 lines Fix a 100% CPU consumption problem when setting + console=yes in asterisk.conf. The handling of -c and console=yes + should be the same, but they were not. When you specify -c, it + sets both a flag for console module and for asterisk not to + fork() off into the background. The handling of console=yes only + set console mode, so you would end up with a background process() + trying to run the Asterisk console and freaking out since it + didn't have anything to read input from. Thanks to beagles for + reporting and helping debug the problem! ........ + +2010-09-22 15:11 +0000 [r288267] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /, UPGRADE.txt: + Merged revisions 288265-288266 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) + | 9 lines Allow the encoding to be set, in case local charset + does not agree with database. (closes issue #16940) Reported by: + jamicque Patches: 20100827__issue16940.diff.txt uploaded by + tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt + uploaded by tilghman (license 14) Tested by: jamicque ........ + r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) + | 5 lines Document addition of encoding parameter. (issue #16940) + Reported by: jamicque ........ + +2010-09-22 00:03 +0000 [r288193] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 288192 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 + Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls + ast_bridged_channel() on an unlocked channel. Near the beginning + of schedule_delivery(), ast_bridged_channel() is called on + iaxs[fr->callno]->owner. However, the channel is not locked, + which can result in ast_bridged_channel() crashing should + owner->tech change to a technology that doesn't implement + bridged_channel. I also fixed the other calls to + ast_bridged_channel() in chan_iax2.c since the owner lock was not + held there either. Converted the existing channel deadlock + avoidance to use iax2_lock_owner(). Using the new function + simplified some awkward code. In the process of fixing the + locking on ast_bridged_channel(), I also found a memory leak in + socket_process() for v1.6.2 and v1.8. The local struct variable + ies.vars is not freed on early/abnormal function exits. (closes + issue #17919) Reported by: rain Patches: issue17919_v1.4.patch + uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch + uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch + uploaded by rmudgett (license 664) Review: + https://reviewboard.asterisk.org/r/926/ ........ + +2010-09-21 22:22 +0000 [r288147] Paul Belanger <paul.belanger@polybeacon.com> + + * channels/chan_iax2.c: Setup timer before set_config(). (closes + issue #18019) Reported by: Netview Patches: issue_0018019.patch + uploaded by pabelanger (license 224) Tested by: Netview + +2010-09-21 21:59 +0000 [r288113] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 288112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) + | 15 lines Try both the encoded and unencoded subscription URI + for a match in hints. When a phone sends an encoded URI for a + subscription, the URI is not matched with the actual hint that is + in decoded format. For example, if we have an extension with a + hint that is named: "#5601" or "*5601", the subscription will + work fine if the phone subscribes with an already decoded URI, + but when it's decoded like "%255601" or "%2A5601", Asterisk is + unable to match it with the correct hint. (closes issue #17785) + Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt + uploaded by tilghman (license 14) Tested by: ramonpeek ........ + +2010-09-21 19:46 +0000 [r288006] Brett Bryant <bbryant@digium.com> + + * main/channel.c, /: Merged revisions 288005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) + | 8 lines Add a check to fix a rare segmentation fault you'd get + if ast_frdup couldn't allocate memory on the first frame being + queued in ast_queue_frame. (closes issue #17882) Reported by: + seanbright Tested by: seanbright ........ + +2010-09-21 19:07 +0000 [r287934] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 287933 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) + | 2 lines Less than zero is an error, not any non-zero value. + ........ + +2010-09-20 23:58 +0000 [r287759] Brett Bryant <bbryant@digium.com> + + * /, apps/app_meetme.c: Merged revisions 287758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) + | 16 lines Fix misvalidation of meetme pins in conjunction with + the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a + user and admin pin setup for your conference, using the user pin + would gain you admin priviledges. Also, when no user pin was set, + an admin pin was, the 'a' MeetMe flag wasn't used, and the user + tried to enter a conference then they were still prompted for a + pin and forced to hit #. (closes issue #17908) Reported by: kuj + Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: + kuj Review: [full review board URL with trailing slash] ........ + +2010-09-20 23:16 +0000 [r287685] Alec L Davis <sivad.a@paradise.net.nz> + + * main/channel.c: ast_channel_masquerade: Avoid recursive + masquerades. Check all 4 combinations of (original/clonechan) * + (masq/masqr). Initially original->masq and clonechan->masqr were + only checked. It's possible with multiple masq's planned - and + not yet executed, that the 'original' chan could already have + another masq'd into it - thus original->masqr would be set, that + masqr would lost. Likewise for the clonechan->masq. (closes issue + #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: + based on bug16057.diff4.txt uploaded by alecdavis (license 585) + Tested by: ramonpeek, davidw, alecdavis + +2010-09-20 21:28 +0000 [r287642] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Don't crash when parking a non-bridged + call. (closes issue #17680) Reported by: jmhunter Patches: + chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: + jmhunter, DEA + 2010-11-02 Leif Madsen <lmadsen@digium.com> * Asterisk 1.6.2.14 Released. |