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2011-01-26Importing release summary for 1.8.3-rc2 release.v1.8.3-rc2lmadsen2-0/+175
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@304140 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merge changes from 303907 into tag.lmadsen3-42/+51
Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@304139 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Remove entry from ChangeLog.lmadsen1-106/+0
The merge for the DTMF based attended transfers was already present in Asterisk 1.8.3-rc1 which is why I didn't merge this last week when RC2 was tagged. git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@303961 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Update ChangeLog and merge in changes for DTMF based attended transfers.lmadsen1-0/+106
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@303959 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Drop these summary files.lmadsen2-152/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@303957 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Importing release summary for 1.8.3-rc2 release.lmadsen2-0/+152
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@303770 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-20Update .version, ChangeLog, and merge changes.lmadsen6-1160/+67
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@303138 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-20Create 1.8.3-rc2 from 1.8.3-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc2@303102 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Use autotagged externalsv1.8.3-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@302179 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Importing release summary for 1.8.3-rc1 release.lmadsen2-0/+1156
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@302177 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Importing files for 1.8.3-rc1 release.lmadsen3-0/+27750
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@302176 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Creating tag for the release of asterisk-1.8.3-rc1lmadsen11-27679/+690
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@302175 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Merged revisions 302173 via svnmerge from rmudgett1-353/+661
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302174 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17Document "encryption" option in sip.conf.sampletwilson1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302005 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Deadlock between dahdi_request() and pri_dchannel() processing an incomming ↵rmudgett1-4/+3
call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301946 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() insteadbbryant1-1/+3
of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301851 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Add relationships to function documentation.lathama2-12/+12
Fix amatuer type mistake git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301849 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Fix for a consistent MulticastRTP channel driver crash due to use of unitilizedbbryant1-1/+1
data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301845 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Add relationships to function documentation.lathama2-0/+20
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301844 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Use autotagged externalslmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@301840 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Importing release summary for 1.8.3-rc1 release.lmadsen2-0/+230
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@301839 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Importing files for 1.8.3-rc1 release.lmadsen3-0/+27092
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@301838 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Creating tag for the release of asterisk-1.8.3-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.3-rc1@301837 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Resolve deadlock involving REFER.jpeeler1-12/+16
Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301790 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-13Merged revisions 301730 via svnmerge from lmadsen1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines Add static entry for split Polycom 332 firmware. (closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301731 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Merged revisions 301682 via svnmerge from twilson1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) | 9 lines Don't reject all SUBSCRIBE auth requests When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301683 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Merged revisions 301594 via svnmerge from mnicholson1-14/+0
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't be necessary in session_do, and removed the ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301595 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Merged revisions 301503 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301504 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Removal of unused variables so Asterisk will compile.dvossel1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301446 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12fix wrong text of rerun menuselect after user interface warningschmidts1-1/+1
the warning, if no user interface for menuselect warning was found is not right. you have to rerun configure before make menuselect after installing a proper user interface. (closes issue #18594) Reported by: Dovid git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301444 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Call execl() directly for a better solution for paths with spaces.tilghman1-5/+3
(closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301402 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-11Merged revisions 301310 via svnmerge from pabelanger1-9/+5
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan 2011) | 2 lines Fix a logic issue when passing context ARG ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301311 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-11Merged revisions 301307 via svnmerge from mnicholson1-11/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600 (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301308 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-10Little endian machines were not converted properly.tilghman1-16/+16
(closes issue #18583) Reported by: jcovert Patches: 20110110__issue18583.diff.txt uploaded by tilghman (license 14) Tested by: jcovert git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301263 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-09Merged revisions 301220 via svnmerge from pabelanger3-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue #18589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301221 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-08Merged revisions 301176 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) | 7 lines Indicate log level argument for Log() is not optional (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301177 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-08The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone.rmudgett1-3/+11
The DAHDI ISDN channel name is not dialable. Make a channel name like DAHDI/i3/400-12 dialable when the sequence number is stripped off of the name. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301134 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-07Merged revisions 301089 via svnmerge from qwell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301090 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-07Merged revisions 301046 via svnmerge from jpeeler1-6/+14
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines Fix regression causing forwarding voicemails to not work with file storage. I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301047 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-07Merged revisions 300951 via svnmerge from jpeeler1-4/+9
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300955 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-06Don't destroy handle not created by use (because the caller will).tilghman1-1/+0
(closes issue #18526) Reported by: makoto Patches: res-config-mysql-include.patch uploaded by makoto (license 38) Tested by: makoto git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300798 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-05Merged revision 300711 fromrmudgett1-3/+55
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines A call retrieved from hold may wind up with no audio. If the retrieved call is natively bridged then the call may not have any audio path. The following warning message is given: "Failed to add <dfd> to conference <chan>/<chan>: Invalid argument". * Open the media on a B channel when pri_fixup_principle() moves the call from a no_b_channel channel to a real channel. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300714 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-05Merged revisions 300622 via svnmerge from tilghman1-8/+9
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600 (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines Use the sanity check in place of the disconnect/connect cycle. The disconnect/connect cycle has the potential to cause random crashes. (closes issue #18243) Reported by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) Tested by: ks3 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300623 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-05Merged revisions 300574 via svnmerge from pabelanger1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan 2011) | 6 lines Change deprecated message to LOG_WARNING Also removed latter part of message Discussed on #asterisk-dev ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300575 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300520 via svnmerge from lmadsen4-6/+6
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300521 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300431 via svnmerge from lmadsen1-1/+28
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011) | 7 lines Add some documentation to users.conf.sample. (closes issue #18531) Reported by: lathama Patches: users.conf.sample2.diff uploaded by lathama (license 1028) Tested by: lathama ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300433 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300429 via svnmerge from russell2-227/+517
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300429 | russell | 2011-01-04 14:59:56 -0600 (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) | 4 lines Update the autosupport script from Digium support. (closes AST-395) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300430 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Update STAT() to use the comma instead of the pipe.lmadsen1-1/+1
(closes issue #18503) Reported by: cjacobsen Patches: old_separator.diff uploaded by cjacobsen (license 1029) Tested by: lathama git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300384 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300298 via svnmerge from twilson1-14/+20
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300301 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Memory leaking in calendarspitel2-0/+2
ne_request_destroy() was missing in icalendar and exchange calendar modules, causing memory leak. (closes issue #18521) Review: https://reviewboard.asterisk.org/r/1068/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300214 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-21 17:49:32 +0000