asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	bd26287e88	Merged revisions 302505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302505 \| seanbright \| 2011-01-19 12:58:11 -0500 (Wed, 19 Jan 2011) \| 14 lines Merged revisions 302504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302504 \| seanbright \| 2011-01-19 12:56:32 -0500 (Wed, 19 Jan 2011) \| 7 lines Make sure that h_length is set when we short-circuit out of ast_gethostbyname. (closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	9d6cbe222c	Merged revisions 302462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302462 \| pabelanger \| 2011-01-19 12:09:35 -0500 (Wed, 19 Jan 2011) \| 9 lines Merged revisions 302461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302461 \| pabelanger \| 2011-01-19 12:08:01 -0500 (Wed, 19 Jan 2011) \| 2 lines Handle 'Resource temporarily unavailable' error more gracefully. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	750dff4f0f	Merged revisions 302417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302417 \| seanbright \| 2011-01-19 10:53:20 -0500 (Wed, 19 Jan 2011) \| 16 lines Merged revisions 302416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302416 \| seanbright \| 2011-01-19 10:52:44 -0500 (Wed, 19 Jan 2011) \| 9 lines Remove references to priorityjumping from the sample extensions.conf. Priority jumping was removed from pbx_config in r68970. (closes issue #18622) Reported by: kshumard Patches: extensions.conf.sample.patch uploaded by kshumard (license 92) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	06ac89965c	Merged revisions 302414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 \| seanbright \| 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) \| 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	d42cb6fd1d	Merged revisions 302412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 \| seanbright \| 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) \| 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8cd1ac534b	Merged revisions 302318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302318 \| rmudgett \| 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) \| 1 line Use the expanded format type instead of plain int. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	785e3a1417	Merged revisions 302314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 \| mnicholson \| 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) \| 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 \| mnicholson \| 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) \| 11 lines Merged revisions 302311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 \| mnicholson \| 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) \| 4 lines URI encode the user part of the contact header. ABE-2705 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b1f9f1e78f	Merged revisions 302266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302266 \| jpeeler \| 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) \| 34 lines Merged revisions 302265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302265 \| jpeeler \| 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) \| 27 lines Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip. Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	519b766cd4	Merged revisions 302267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302267 \| russell \| 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) \| 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on. AO2_DEBUG is not important and is causing a false compiler warning to be generated on my Ubuntu Natty dev box. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a05aeff312	Merged revisions 302174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302174 \| rmudgett \| 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) \| 102 lines Merged revisions 302173 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@302178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	ae6b55e4a3	Merged revisions 293493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293493 \| twilson \| 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) \| 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	29cb03ebf2	Merged revisions 302005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302005 \| twilson \| 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) \| 2 lines Document "encryption" option in sip.conf.sample ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c69406f384	Merged revisions 301946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301946 \| rmudgett \| 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) \| 13 lines Deadlock between dahdi_request() and pri_dchannel() processing an incomming 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: https://origsvn.digium.com/svn/asterisk/trunk@301947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ed0a2e8c31	Merged revisions 301851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301851 \| bbryant \| 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) \| 6 lines Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	7cb1c06dd3	Add relationships to function documentation. Fix amatuer type mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	558c6a5a1a	Merged revisions 301845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301845 \| bbryant \| 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) \| 9 lines Fix for a consistent MulticastRTP channel driver crash due to use of unitilized data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	ca8a5498b1	Add relationships to function documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a0e4c4ee5b	Merged revisions 301790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 \| jpeeler \| 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) \| 42 lines Resolve deadlock involving REFER. 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: https://origsvn.digium.com/svn/asterisk/trunk@301791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	89fe21382a	Merged revisions 301731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301731 \| lmadsen \| 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) \| 15 lines Merged revisions 301730 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	f485bfd1d3	Add dialplan variables for asterisk.conf directories Review: https://reviewboard.asterisk.org/r/1075/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c6858b9a1d	Merged revisions 301683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 \| twilson \| 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) \| 15 lines Merged revisions 301682 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	8ad7304e66	Merged revisions 301595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301595 \| mnicholson \| 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) \| 22 lines Merged revisions 301594 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a307b5407e	Merged revisions 301504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301504 \| jpeeler \| 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) \| 26 lines Merged revisions 301503 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2a618dc998	Merged revisions 301446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301446 \| dvossel \| 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) \| 2 lines Removal of unused variables so Asterisk will compile. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Stefan Schmidt	f6834d3247	fix wrong text of rerun menuselect after user interface warning 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 0018594) Reported by: Dovid git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fad87eea35	Merged revisions 301402 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301402 \| tilghman \| 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) \| 7 lines Call execl() directly for a better solution for paths with spaces. (closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	ca8d5676ab	Merged revisions 301311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301311 \| pabelanger \| 2011-01-11 14:16:06 -0500 (Tue, 11 Jan 2011) \| 9 lines Merged revisions 301310 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	50a0c8a646	Merged revisions 301308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301308 \| mnicholson \| 2011-01-11 12:51:40 -0600 (Tue, 11 Jan 2011) \| 18 lines Merged revisions 301307 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	cbf80fd534	Merged revisions 301263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301263 \| tilghman \| 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011) \| 8 lines Little endian machines were not converted properly. (closes issue #18583) Reported by: jcovert Patches: 20110110__issue18583.diff.txt uploaded by tilghman (license 14) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5fc47953f7	Merged revisions 301221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301221 \| pabelanger \| 2011-01-09 16:40:34 -0500 (Sun, 09 Jan 2011) \| 21 lines Merged revisions 301220 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	563d973c11	Merged revisions 301177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301177 \| pabelanger \| 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) \| 14 lines Merged revisions 301176 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f91340bb71	Merged revisions 301134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301134 \| rmudgett \| 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) \| 7 lines The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone. 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: https://origsvn.digium.com/svn/asterisk/trunk@301135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	74e0a87776	Merged revisions 301090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301090 \| qwell \| 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) \| 15 lines Merged revisions 301089 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	ac11bca7c0	Merged revisions 301047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301047 \| jpeeler \| 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) \| 15 lines Merged revisions 301046 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@301048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b2a70b4065	Oops, missed the actual decoding part. (closes issue #18046) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	3eec341083	Merged revisions 300955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300955 \| jpeeler \| 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) \| 21 lines Merged revisions 300951 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Murawki	a7f9ce2e77	Added support for postgres database retry query on disconnection to res_config_pgsql If your postgres connection died suddenly in between res_config_pgsql queries, the next query will fail because the query is executed on a disconnected/disconnecting handle. The query is abandoned and is returned from in error. Now we will reconnect and try again if a query was run on a disconnected connection. (closes issue #18071) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a58b2fb395	XML validation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	473e176df8	Add a hashcompat mode called "legacy", which translates a literal plus sign to a space. (closes issue #18046) Reported by: wdoekes Patches: 20100930__issue18046.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a1aa18b8ac	Merged revisions 300798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r300798 \| tilghman \| 2011-01-06 00:28:18 -0600 (Thu, 06 Jan 2011) \| 8 lines Don't destroy handle not created by use (because the caller will). (closes issue #18526) Reported by: makoto Patches: res-config-mysql-include.patch uploaded by makoto (license 38) Tested by: makoto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Ruggles	08e6f86d23	update safe_asterisk script change defaults to make a little more sense. Default log location is now asterisk log location and default email notification has been changed to root on the local machine Review: https://reviewboard.asterisk.org/r/1067/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	398d633ce0	Merged revisions 300714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300714 \| rmudgett \| 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) \| 21 lines Merged revision 300711 from 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: https://origsvn.digium.com/svn/asterisk/trunk@300716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6f5c681843	Merged revisions 300623 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300623 \| tilghman \| 2011-01-05 12:56:12 -0600 (Wed, 05 Jan 2011) \| 24 lines Merged revisions 300622 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	8b23fd448c	Merged revisions 300575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300575 \| pabelanger \| 2011-01-05 11:29:19 -0500 (Wed, 05 Jan 2011) \| 13 lines Merged revisions 300574 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	783ea39ba1	Merged revisions 300521 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 \| lmadsen \| 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) \| 17 lines Merged revisions 300520 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b9271a15e5	Merged revisions 300433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300433 \| lmadsen \| 2011-01-04 15:00:55 -0600 (Tue, 04 Jan 2011) \| 15 lines Merged revisions 300431 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	eb9f9bcba6	Merged revisions 300430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300430 \| russell \| 2011-01-04 15:00:16 -0600 (Tue, 04 Jan 2011) \| 18 lines Merged revisions 300429 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	e06e5b2e81	Merged revisions 300384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r300384 \| lmadsen \| 2011-01-04 13:45:22 -0600 (Tue, 04 Jan 2011) \| 7 lines Update STAT() to use the comma instead of the pipe. (closes issue #18503) Reported by: cjacobsen Patches: old_separator.diff uploaded by cjacobsen (license 1029) Tested by: lathama ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Moises Silva	3b1553f281	Update MFC-R2 code to use new DTMF-R2 functionality in OpenR2 (closes issue #18576) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	94ef793caa	Merged revisions 300301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300301 \| twilson \| 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) \| 29 lines Merged revisions 300298 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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