2009-07-27 Leif Madsen * Asterisk 1.6.1.2 released 2009-06-05 Leif Madsen * Asterisk 1.6.1.1 released 2009-06-04 David Vossel * channels/chan_iax2.c: Additional updates for AST-2009-001 2009-06-04 David Vossel * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 2009-04-27 Leif Madsen * Create Asterisk 1.6.1.0 2009-04-20 Leif Madsen * Create Asterisk 1.6.1.0-rc5 2009-04-20 17:08 +0000 [r189352] Joshua Colp * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines Fix a bug with non-UDP connections that caused dialogs to not get freed. This issue crept up because of a reference count issue on non-UDP based dialogs. The dialog reference count was increased when transmitting a packet reliably but never decreased. This caused the dialog structure to hang around despite being unlinked from the dialogs container. (closes issue #14919) Reported by: vrban ........ 2009-04-20 14:06 +0000 [r189280] Mark Michelson * main/channel.c, /: Merged revisions 189278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr 2009) | 18 lines Merged revisions 189277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines Move the check for chan->fdno == -1 to after the zombie/hangup check. Many users were finding that their hung up channels were staying up and causing 100% CPU usage. (issue #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch uploaded by mmichelson (license 60) Tested by: falves11, bamby ........ ................ 2009-04-18 01:38 +0000 [r189206] David Vossel * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ ................ 2009-04-17 21:55 +0000 [r189139] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 189137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) | 17 lines Merged revisions 188833,189134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ ................ 2009-04-17 20:21 +0000 [r189103] Mark Michelson * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines Prevent a crash when SIP blonde transferring an unbridged call. If one attempts to use the attended transfer button on a SIP phone to transfer an unbridged call (such as a call to an IVR) but hangs up while the target of the transfer is still ringing, we need to not crash. The problem was that ast_hangup was called from outside the channel thread. AST-211 ........ 2009-04-17 19:46 +0000 [r189080] Sean Bright * main/asterisk.c, /: Merged revisions 189077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 line Fix copy/paste error with 'transmit silence' flag. ........ 2009-04-17 17:33 +0000 [r189069] Matthew Nicholson * main/pbx.c, /: Merged revisions 189010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr 2009) | 12 lines Merged revisions 189009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr 2009) | 5 lines Make Busy() application set the CDR disposition to BUSY. (closes issue #14306) Reported by: cristiandimache ........ ................ 2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines Merged revisions 188946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines Fix a bug where a value used to create the channel name was bogus. This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file ........ ................ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been. (issue AST-210) ........ ................ 2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) | 14 lines Merged revisions 188835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines Only update realtime, if global option rtupdate != false (closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh ........ ................ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500 (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) | 4 lines Umask should not be exported into global namespace. (closes issue #14912) Reported by: jcapp ........ ................ 2009-04-15 22:12 +0000 [r188649] David Vossel * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines National prefix inserted even when caller ID not available When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ ........ ................ 2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson * /, main/file.c: Merged revisions 188585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr 2009) | 13 lines Merged revisions 188582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines Update ast_readvideo_callback to match ast_readaudio_callback. This fixes potential refcount errors that may occur on ast_filestreams. AST-208 ........ ................ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 | mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 lines Fix a couple of queue member reference leaks. ........ 2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp * main/rtp.c, /: Merged revisions 188413 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix an incorrect clock rate when sending T140 text. (closes issue #14029) Reported by: epicac ........ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix a bug with the change I made yesterday to outbound proxy support. Per discussion with oej on IRC we need the actual IP address, not the outbound proxy IP address, in the sa field. Upon further inspection this should make the behaviour of all other uses of the outbound proxy in the code. ........ 2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher * main/pbx.c, /: Merged revisions 188210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 | tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas. ........ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 | tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines Application delimiter is ',', not '|'. (closes issue #14881) Reported by: stegro Patches: smsq.patch uploaded by stegro (license 752) ........ 2009-04-13 19:33 +0000 [r188104] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr 2009) | 5 lines Fix another crash related to cached realtime music on hold. This was another off-by-one problem caused by moh_register. ........ 2009-04-13 16:32 +0000 [r188069] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1. Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will be sending to. This has to be done because the logic that determines what local IP address to use in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address we are sending to. (closes issue #12006) Reported by: mnicholson ........ 2009-04-13 14:20 +0000 [r188038] Mark Michelson * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines Set all queue variables on both the caller and member channels. This allows for the variables to be accessed if a member macro is run. Thanks to Grigoriy Puzankin for bringing this up on the -dev list. ........ 2009-04-10 20:28 +0000 [r187914] Jeff Peeler * channels/Makefile, /: Merged revisions 187906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines Fix module embedding for chan_h323. Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome Patches: issue_11966.patch uploaded by kpfleming (license 421) Tested by: jpeeler ........ 2009-04-10 17:30 +0000 [r187767] Tilghman Lesher * contrib/scripts/sip-friends.sql, contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500 (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines Add lastms column to the contributed table designs ........ ................ 2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming * /, build_tools/embed_modules.xml: Merged revisions 187721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 Apr 2009) | 5 lines clean up some patterns for files to remove add embedding support for bridge and test modules ........ 2009-04-10 16:03 +0000 [r187678] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj ........ 2009-04-10 16:00 +0000 [r187676] Russell Bryant * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines Disable test modules by default. ........ 2009-04-10 03:56 +0000 [r187600] Tilghman Lesher * main/channel.c, main/pbx.c, main/manager.c, /, include/asterisk/linkedlists.h, main/features.c, main/http.c, main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged revisions 187599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines Modify headers and macros, according to Russell's suggestions on the -dev list ........ 2009-04-09 19:14 +0000 [r187495] Mark Michelson * /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr 2009) | 24 lines Merged revisions 187484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines Handle a SIP race condition (reinvite before an ACK) properly. RFC 5047 explains the proper course of action to take if a reINVITE is received before the ACK from a previous invite transaction. What we are to do is to treat the reINVITE as if it were both an ACK and a reINVITE and process it normally. Later, when we receive the ACK we had been expecting, we will ignore it since its CSeq is less than the current iseqno of the sip_pvt representing this dialog. (closes issue #13849) Reported by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson (license 60) Tested by: mmichelson, klaus3000 ........ ................ 2009-04-09 18:54 +0000 [r187486] Tilghman Lesher * main/manager.c, /, include/asterisk/linkedlists.h, include/asterisk/lock.h: Merged revisions 187483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines Race condition between ast_cli_command() and 'module unload' could cause a deadlock. Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan ........ ................ 2009-04-09 17:43 +0000 [r187427] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 187421,187424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using cached realtime moh. The moh_register function links an mohclass and then immediately unrefs the class since the container now has a reference. The problem with using realtime music on hold is that the class is allocated, registered, and started in one fell swoop. The refcounting logic resulted in the count being off by one. The same problem did not happen when using a static config because the allocation and registration of an mohclass is a separate operation from starting moh. This also did not affect non-cached realtime moh because the classes are not registered at all. I also have modified res_musiconhold to use the _t_ variants of the ao2_ functions so that more info can be gleaned when attempting to trace the refcounts. I found this to be incredibly helpful for debugging this issue and there's no good reason to remove it. (closes issue #14661) Reported by: sum ........ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines Use safe macro practices even though they really aren't necessary. ........ 2009-04-09 17:22 +0000 [r187305-187388] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines Allow '/' in username portion of register; this is a regression. (closes issue #14668) Reported by: Netview ........ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 187363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ ................ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, utils/Makefile, include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c (added): Merged revisions 187302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ ................ 2009-04-08 16:53 +0000 [r186987-187048] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines Fix a small logical error when loading moh classes. We were unconditionally incrementing the number of mohclasses registered. However, we should actually only increment if the call to moh_register was successful. While this probably has never caused problems, I noticed it and decided to fix it anyway. ........ ................ * main/channel.c, /: Merged revisions 186985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines Merged revisions 186984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines Make a couple of changes with regards to a new message printed in ast_read(). "ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. ........ ................ 2009-04-08 05:07 +0000 [r186900] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines Add lastms to the require API call. ........ 2009-04-08 00:10 +0000 [r186835-186844] Mark Michelson * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged revisions 186842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines Merged revisions 186841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ ................ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ * main/channel.c, /: Merged revisions 186833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ ................ 2009-04-07 22:33 +0000 [r186806] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines Merged revisions 186775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines Fix Macro documentation to match current (and intended) behavior. (See -dev mailing list) ........ ................ 2009-04-07 20:53 +0000 [r186722] Mark Michelson * main/manager.c, /: Merged revisions 186720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines Merged revisions 186719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines Ensure that \r\n is printed after the ActionID in an OriginateResponse. (closes issue #14847) Reported by: kobaz ........ ................ 2009-04-03 20:21 +0000 [r186466] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ 2009-04-03 20:04 +0000 [r186448] Tilghman Lesher * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 186444,186447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines Found a conflict in the last commit, due to multiple targets ........ ................ 2009-04-03 16:38 +0000 [r186381] David Vossel * /, main/audiohook.c: Merged revisions 186379 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines audio_audiohook_write_list() did not correctly update sample size after ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) ........ 2009-04-03 Leif Madsen * Asterisk 1.6.1.0-rc4 released. 2009-04-03 15:54 +0000 [r186323] Joshua Colp * include/asterisk/crypto.h, /: Merged revisions 186321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ 2009-04-03 14:33 +0000 [r186288] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines Fix the ability to retrieve voicemail messages from IMAP. A recent change made interactive vm_states no longer get added to the list of vm_states and instead get stored in thread-local storage. In trunk and all the 1.6.X branches, the problem is that when we search for messages in a voicemail box, we would attempt to update the appropriate vm_state struct by directly searching in the list of vm_states instead of using the get_vm_state_by_imap_user function. This meant we could not find the interactive vm_state that we wanted. (closes issue #14685) Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ 2009-04-03 02:06 +0000 [r186232] Russell Bryant * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines Merged revisions 186229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines Fix a memory leak in cdr_radius. I came across this while doing some testing of my ast_channel_ao2 branch. After running a test overnight that generated over 5 million calls, Asterisk had taken up about 1 GB of my system memory. So, I re-ran the test with MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the test, even though Asterisk was still consuming it somehow. Instead, I turned to valgrind, which when run with --leak-check=full, told me exactly where the leak came from, which was from allocations inside the radiusclient-ng library. This explains why MALLOC_DEBUG did not report it. After a bit of analysis, I found that we were leaking a little bit of memory every time a CDR record was passed to cdr_radius. I don't actually have a radius server set up to receive CDR records. However, I always have my development systems compile and install all modules. In addition to making sure there are not build errors across modules, always loading modules helps find bugs like this, too, so it is strongly recommend for all developers. ........ ................ 2009-04-02 21:59 +0000 [r186177] Mark Michelson * configs/features.conf.sample, /: Merged revisions 186175 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ 2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ 2009-04-02 17:14 +0000 [r186062] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 186060 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ 2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ 2009-04-01 19:06 +0000 [r185848] David Vossel * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ 2009-04-01 13:50 +0000 [r185774] Russell Bryant * main/channel.c, /: Merged revisions 185772 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ ................ 2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming * utils, /: Merged revisions 185664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line ignore copied (generated) file ........ 2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar 2009) | 12 lines Merged revisions 185599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines Fix crash that would occur if an empty member was specified in queues.conf. (closes issue #14796) Reported by: pida ........ ................ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the word "messages" properly. (closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) ........ ................ 2009-03-31 17:48 +0000 [r185427] David Brooks * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: Review: http://reviewboard.digium.com/r/181/ ........ ................ 2009-03-31 14:57 +0000 [r185263] Russell Bryant * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto ........ 2009-03-31 14:10 +0000 [r185199] Joshua Colp * /, main/audiohook.c: Merged revisions 185197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines Merged revisions 185196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ ................ 2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged revisions 185123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ 2009-03-30 16:47 +0000 [r185088] Mark Michelson * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ ................ 2009-03-30 14:41 +0000 [r184950] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ 2009-03-30 13:57 +0000 [r184912] Russell Bryant * channels/h323/Makefile.in, /: Merged revisions 184910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ 2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ ................ * channels/Makefile, /: Merged revisions 184838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ 2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming * channels/chan_iax2.c, main/timing.c, main/channel.c, /, include/asterisk/timing.h, include/asterisk/channel.h: Merged revisions 184762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ 2009-03-27 18:09 +0000 [r184728] Russell Bryant * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ 2009-03-27 15:54 +0000 [r184675] Joshua Colp * /, res/res_agi.c: Merged revisions 184673 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix speech structure leak in the AGI speech recognition integration. The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) ........ 2009-03-27 14:04 +0000 [r184631] Russell Bryant * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /, res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions 184630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines Change g_eid to ast_eid_default. ........ 2009-03-27 13:22 +0000 [r184587] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ 2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009) | 20 lines Fix some issues with rwlock corruption that caused deadlock like symptoms. When dvossel and I were doing some load testing last week, we noticed that we could make Asterisk trunk lock up instantly when we started generating a bunch of calls. The backtraces of locked threads were bizarre, and many were stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a number of places where a backtrace would be loaded into an invalid index of the backtrace array. It's an off by one error, which ends up writing over the rwlock itself. 2) Ensure that in the array of held locks, we NULL out an index once it is not being used so that it's not confusing when analyzing its contents. 3) Remove a bunch of logging referring to an rwlock operating being done with "deep reentrancy". It is normal for _many_ threads to hold a read lock on an rwlock. ........ * /, main/file.c: Merged revisions 184515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines Don't act surprised if we get a -1 indication. ........ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26 Mar 2009) | 2 lines Pass more useful information through to lock tracking when DEBUG_THREADS is on. ........ 2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 184448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar 2009) | 9 lines Merged revisions 184447 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines use new, improved 8kHz prompts ........ ................ 2009-03-26 21:18 +0000 [r184394] David Vossel * /, apps/app_test.c: Merged revisions 184389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009) | 14 lines Merged revisions 184388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF 8 app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir ........ ................ 2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant * /, main/event.c: Merged revisions 184344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 | russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines Remove unneeded AST_LIST_ENTRY() and comment on the purpose of ast_event_ref. ........ * channels/chan_iax2.c, channels/chan_dahdi.c, include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c, main/event.c, include/asterisk/strings.h, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c, channels/chan_unistim.c, include/asterisk/devicestate.h, /, channels/chan_sip.c, main/devicestate.c, include/asterisk/_private.h: Merged revisions 184339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ * /: add reviewboard:url property. 2009-03-25 19:26 +0000 [r184282] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix issue with a T38 reinvite being sent even if not configured to do so. If we receive a T38 request negotiate control frame we should only attempt to do so if the option is enabled on the dialog. ........ 2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons * main/asterisk.c, /: Merged revisions 184220 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines Merged revisions 184188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen ........ ................ 2009-03-25 01:55 +0000 [r184149] Russell Bryant * main/timing.c, utils/Makefile, /, include/asterisk/compat.h: Merged revisions 184147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ 2009-03-24 22:42 +0000 [r184081] Mark Michelson * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines Merged revisions 184078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ ................ 2009-03-24 21:47 +0000 [r184039] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low and =medium The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. ........ 2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher * /, configs/voicemail.conf.sample: Merged revisions 183914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ * /, main/http.c: Merged revisions 183865 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines Allow browsers to cache images and other static content. (This is a regression over 1.4) ........ 2009-03-23 18:59 +0000 [r183768] Mark Michelson * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines Merged revisions 183700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. ........ ................ 2009-03-23 18:12 +0000 [r183703] Leif Madsen * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen ........ 2009-03-20 17:08 +0000 [r183563] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ ................ 2009-03-19 20:33 +0000 [r183438] David Vossel * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: Merged revisions 183436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ 2009-03-19 19:19 +0000 [r183333] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-03-19 18:14 +0000 [r183249] Russell Bryant * main/loader.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 183242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ 2009-03-19 18:11 +0000 [r183246] Mark Michelson * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ 2009-03-19 17:08 +0000 [r183198] David Vossel * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: Merged revisions 183172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ 2009-03-19 16:09 +0000 [r183121] Mark Michelson * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ 2009-03-19 Leif Madsen * Release Asterisk 1.6.1.0-rc3 2009-03-19 15:43 +0000 [r183067-183110] Joshua Colp * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ * main/channel.c, /: Merged revisions 183057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ 2009-03-18 21:19 +0000 [r183030] Jeff Peeler * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ 2009-03-18 14:32 +0000 [r182946] Russell Bryant * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c, include/asterisk/poll-compat.h, channels/chan_alsa.c, main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ 2009-03-17 20:52 +0000 [r182724] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx, configure, autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h, channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions 182722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ 2009-03-17 15:31 +0000 [r182570] Russell Bryant * main/channel.c, /: Merged revisions 182553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ 2009-03-17 15:00 +0000 [r182527-182533] Kevin P. Fleming * main/channel.c, /: Merged revisions 182530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 182525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ 2009-03-17 05:54 +0000 [r182452] Tilghman Lesher * main/db.c, /: Merged revisions 182450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ 2009-03-16 17:53 +0000 [r182284] David Vossel * channels/chan_iax2.c, /: Merged revisions 182282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ 2009-03-16 17:38 +0000 [r182280] Tilghman Lesher * channels/chan_local.c, /, funcs/func_env.c: Merged revisions 182211,182278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC ................ 2009-03-16 14:00 +0000 [r182173] Joshua Colp * main/channel.c, /: Merged revisions 182171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. ........ 2009-03-13 21:27 +0000 [r182068-182123] Mark Michelson * apps/app_queue.c, /: Merged revisions 182121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy ........ * /, main/features.c: Merged revisions 182029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ 2009-03-13 17:29 +0000 [r182042] Joshua Colp * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ 2009-03-13 16:58 +0000 [r181987] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 181985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line improve a bit of suboptimal code ........ 2009-03-12 21:45 +0000 [r181771-181849] Mark Michelson * apps/app_queue.c, /: Merged revisions 181846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines Run the macro on the queue member's channel when he answers, not the caller's channel. ........ * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ 2009-03-12 18:07 +0000 [r181733] Tilghman Lesher * /, main/translate.c: Merged revisions 181731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj ........ 2009-03-12 16:58 +0000 [r181614-181667] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ ................ * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ 2009-03-12 01:05 +0000 [r181544] Richard Mudgett * /, build_tools/make_version: Merged revisions 181542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line Use the correct branch integrated property when generating the version string ........ 2009-03-11 23:21 +0000 [r181521] Michiel van Baak * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk 2009-03-11 22:27 +0000 [r181474] Russell Bryant * main/channel.c, /: Merged revisions 181465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. ........ 2009-03-11 22:23 +0000 [r181457] Jason Parker * /, configure, configure.ac: Merged revisions 181444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ ................ 2009-03-11 22:16 +0000 [r181426-181430] Russell Bryant * main/channel.c, /: Merged revisions 181428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ * main/channel.c, /: Merged revisions 181424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ 2009-03-11 17:40 +0000 [r181373] David Vossel * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions 181371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ 2009-03-11 17:29 +0000 [r181298-181359] Joshua Colp * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ 2009-03-11 15:54 +0000 [r181199-181283] Jeff Peeler * channels/h323/ast_h323.cxx: add missing header file * pbx/pbx_config.c, utils/Makefile, include/asterisk/utils.h, include/asterisk/astmm.h, /, channels/chan_sip.c, channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c: Merged revisions 181135 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ 2009-03-11 01:04 +0000 [r181035] Mark Michelson * /, channels/chan_sip.c: Merged revisions 181032-181033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ 2009-03-10 22:07 +0000 [r180947] Jason Parker * /, configure, configure.ac, autoconf/ast_prog_sed.m4, autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ ................ 2009-03-10 14:42 +0000 [r180802] Joshua Colp * main/manager.c, /: Merged revisions 180800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ 2009-03-09 21:22 +0000 [r180740] Jeff Peeler * include/asterisk/heap.h, include/asterisk/http.h, include/asterisk/logger.h, main/tcptls.c, include/asterisk/res_odbc.h, include/asterisk/doxyref.h, include/asterisk/event.h, include/asterisk/audiohook.h, include/asterisk/dsp.h, include/asterisk/lock.h, include/asterisk/udptl.h, include/asterisk/dnsmgr.h, include/asterisk/utils.h, include/asterisk/devicestate.h, /, include/asterisk/taskprocessor.h, include/asterisk/astobj2.h, include/asterisk/channel.h, include/asterisk/tcptls.h, include/asterisk/manager.h, main/enum.c, include/asterisk/callerid.h, include/asterisk/app.h, include/asterisk/linkedlists.h, include/asterisk/sched.h, include/asterisk/datastore.h, include/asterisk/timing.h, include/asterisk/dlinkedlists.h, include/asterisk/pbx.h, include/asterisk/enum.h, include/asterisk/config.h, include/asterisk/rtp.h, include/asterisk/extconf.h, main/devicestate.c: Merged revisions 180719 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ ........ 2009-03-06 18:26 +0000 [r180585] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ ................ 2009-03-06 17:35 +0000 [r180537] David Vossel * main/enum.c, /: Merged revisions 180534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ ................ 2009-03-05 23:28 +0000 [r180425-180467] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 180383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ 2009-03-05 18:40 +0000 [r180378] Kevin P. Fleming * include/asterisk/frame.h, main/rtp.c, main/frame.c, /: Merged revisions 180373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ 2009-03-04 Leif Madsen * Released Asterisk 1.6.1.0-rc2 2009-03-04 21:09 +0000 [r180263] Russell Bryant * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ ........ 2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp * /, main/callerid.c: Merged revisions 180195 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ ................ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) ........ 2009-03-03 23:39 +0000 [r180080] David Vossel * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, main/app.c: Merged revisions 180032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ 2009-03-03 23:31 +0000 [r180077] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Merged revisions 179973 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ ................ 2009-03-03 22:49 +0000 [r179939-180009] Mark Michelson * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions 180007 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged revisions 179937 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ ........ 2009-03-03 20:09 +0000 [r179905] Russell Bryant * /, apps/app_directed_pickup.c: Merged revisions 179903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line fix a leaked channel lock (and future deadlock) when we try to pick up our own channel ........ 2009-03-03 18:30 +0000 [r179843] Joshua Colp * /, main/features.c: Merged revisions 179841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ 2009-03-03 16:48 +0000 [r179744] Russell Bryant * main/channel.c, /: Merged revisions 179742 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ 2009-03-03 14:41 +0000 [r179674] Joshua Colp * main/channel.c, /: Merged revisions 179672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ 2009-03-03 13:56 +0000 [r179611] Russell Bryant * main/channel.c, /: Merged revisions 179609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ 2009-03-03 00:04 +0000 [r179539] Jeff Peeler * main/channel.c, /: Merged revisions 179537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ 2009-03-02 23:39 +0000 [r179535] Russell Bryant * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ 2009-03-02 23:12 +0000 [r179471] Tilghman Lesher * /, main/app.c: Merged revisions 179469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ ................ 2009-03-02 23:04 +0000 [r179464] Russell Bryant * main/channel.c, /: Merged revisions 179462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ 2009-03-02 20:18 +0000 [r179407] Jason Parker * /, main/editline/configure, main/editline/np/unvis.c, main/editline/sys.h, main/editline/configure.in: Merged revisions 179396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ ................ 2009-03-02 17:19 +0000 [r179362] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009) | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not loaded) ........ 2009-03-02 14:14 +0000 [r179293] Joshua Colp * /, main/audiohook.c: Merged revisions 179291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) ........ 2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........ * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ 2009-02-27 21:48 +0000 [r179166] Russell Bryant * configs/ais.conf.sample, res/res_ais.c, /, doc/distributed_devstate.txt: Merged revisions 179164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines Mark res_ais as experimental, as the binary event format is subject to change. ........ 2009-02-27 21:34 +0000 [r179163] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines If config file is blank, don't load module. (Closes issue #14563) ........ 2009-02-27 21:25 +0000 [r179160] Russell Bryant * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines Add a note about the ordering of entries in sip.conf in 1.6.1. ........ 2009-02-27 19:06 +0000 [r179059] Jason Parker * /, doc/tex/channelvariables.tex: Merged revisions 179057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) ........ 2009-02-27 03:56 +0000 [r178988] Steve Murphy * configs/features.conf.sample, /, main/features.c: Merged revisions 178986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ 2009-02-26 17:50 +0000 [r178875] David Vossel * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ 2009-02-26 17:38 +0000 [r178869] Steve Murphy * /, main/features.c: Merged revisions 178828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ 2009-02-26 16:44 +0000 [r178803] Joshua Colp * /, main/file.c: Merged revisions 178801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant ........ 2009-02-26 16:07 +0000 [r178769] David Vossel * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ 2009-02-25 12:46 +0000 [r178511] Russell Bryant * main/asterisk.c, /: Merged revisions 178509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ ................ 2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 178446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines Apparently, a void cast doesn't override warn_unused_result. ........ 2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant * Makefile: revert accidental Makefile change. * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ ................ 2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher * main/asterisk.c, /: Merged revisions 178375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines The 3 possible errors with pipe(2) are all impossible in this situation. ........ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. ........ * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. Also, add some documentation supporting the use of astcanary. (closes issue #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) ........ 2009-02-24 15:22 +0000 [r178232] Joshua Colp * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ ................ 2009-02-23 23:22 +0000 [r178172] Russell Bryant * main/rtp.c, /: Merged revisions 178142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-21 16:04 +0000 [r177945] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 | tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines On update, test against the existence of sipregs. ........ 2009-02-21 12:51 +0000 [r177851] Michiel van Baak * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 | mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines make chan_sip.c compile on OpenBSD again. ........ 2009-02-20 23:05 +0000 [r177789] Tilghman Lesher * main/pbx.c, /: Merged revisions 177787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines Merged revisions 177786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines Don't print the CR-NL combination when we aren't outputting to the manager. An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman ........ ................ 2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 | dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines Make app_fax compatible with spandsp-0.0.6pre4 Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. ........ 2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged revisions 177664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ * include/asterisk/threadstorage.h, /: Merged revisions 177732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ ................ 2009-02-20 20:34 +0000 [r177700] David Vossel * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert 2009-02-20 17:28 +0000 [r177663] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 177661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) | 2 lines Oops, merge broke trunk ........ 2009-02-20 00:38 +0000 [r177626] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 177624 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177624 | jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines Set sip_request ast_str data to NULL so ast_str_copy allocates space properly in copy_request (issue #14478) Reported by: erik_dedecker ........ 2009-02-20 00:26 +0000 [r177623] Steve Murphy * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was already pretty 8-bit clean; but I'm still removing the --full from the flex command so everything is uniform. ........ r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines This patch fixes a problem with 8-bit input to the ast_expr2 scanner. The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf ........ ................ 2009-02-19 22:35 +0000 [r177539] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 177537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177537 | tilghman | 2009-02-19 16:33:00 -0600 (Thu, 19 Feb 2009) | 14 lines Merged revisions 177536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin ........ ................ 2009-02-19 16:46 +0000 [r177389] Jeff Peeler * /, include/asterisk/channel.h: Merged revisions 177387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines Fix another merge error from 176708 ........ 2009-02-19 16:40 +0000 [r177386] Joshua Colp * apps/app_speech_utils.c, /: Merged revisions 177384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ ................ 2009-02-19 15:57 +0000 [r177358] Jeff Peeler * /, main/features.c: Merged revisions 177356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! ........ 2009-02-19 00:17 +0000 [r177294] Steve Murphy * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed, 18 Feb 2009) | 39 lines Merged revisions 177225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines This patch fixes a regression of sorts that was introduced in rev 24425. It basically fixes AST-190/ABE-1782. What was wrong: the user has 6000 extensions in one context; and then 6000 contexts, one per extension. The parser could only handle about 4893 of the 6000 extens in the single context. This was due to the regression I mentioned. To get rid of shift/reduce conflicts, Luigi set up right-recursive lists for globals, context elements, switch lists, and statements. Right recursive lists got rid of the warnings, but instead, they use up a tremendous amount of stack space when the lists are long. I saw this a few years back, and resolved not to fix it until someone complained. That day has arrived! After the changes were made, I ran the regression test suite, and there were no problems. I took the test case the user provided, and added 100,000 extensions to the single context, that already had 6,000 extens in it. (I'll see your 6, and raise you 100!) It takes a few minutes to read it all in, check it and generate code for it, but no problems. So, I think I can say that fundamentally, there are no longer any limits on the number of items you can place in contexts, statement blocks, switches, or globals, beyond your virt mem constraints. ........ ................ 2009-02-18 23:15 +0000 [r177230] Kevin P. Fleming * main/frame.c, /: Merged revisions 177229 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 | kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps. ........ 2009-02-18 23:03 +0000 [r177228] David Vossel * /, main/features.c: Merged revisions 177226 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ ........ 2009-02-18 20:16 +0000 [r177164] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h, /, channels/h323/caps_h323.cxx, channels/h323/ast_ptlib.h (added), channels/h323/ast_h323.cxx, configure, channels/h323/compat_h323.h, configure.ac, channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, channels/h323/ast_h323.h: Merged revisions 177162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) | 14 lines Modify h323 to build against PTLib as well as the older PWLib Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler ........ 2009-02-18 19:30 +0000 [r177158] Russell Bryant * /, apps/app_meetme.c: Merged revisions 177101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 | russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave ........ 2009-02-18 19:09 +0000 [r177100] Tilghman Lesher * /, include/asterisk/config.h: Merged revisions 177098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600 (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines Document the return value of the update method (as requested on -dev list) ........ ................ 2009-02-18 17:26 +0000 [r177037] Doug Bailey * /, main/utils.c: Merged revisions 177035 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines Fixed error where a check for an zero length, terminated string was needed. ........ 2009-02-18 17:14 +0000 [r177007] Joshua Colp * /, channels/chan_sip.c: Merged revisions 177005 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix ordering of output for a ChannelUpdate manager event. (closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) ........ 2009-02-18 16:20 +0000 [r176962] Doug Bailey * /, main/utils.c: Merged revisions 176948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 | dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines Need to take into account the \0 terminator of the old string to determine the amount available. ........ 2009-02-18 15:59 +0000 [r176946] Steve Murphy * main/pbx.c, /: Merged revisions 176943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 | murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present. Reason: when I re-engineered the merge_and_delete func to reduce its lock time, I failed to notice that the functions it calls still also do locking as before. This leads to deadlocks on dialplan reloads, when there are actually living, subscribed hints registered in the system. While the reporter come across this problem while using AEL, I might note that these deadlocks should also happen if extensions.conf were used. Here I added these routines to pbx.c: ast_add_extension_nolock add_pri_lockopt ast_add_extension2_lockopt find_context add_hint_nolock All of the above routines are static and restricted to be used only within pbx.c, and more specifically within the merge_contexts_and_delete routine. They are pretty much the same as their counterparts except they don't lock contexts or hints. Most of them now do the real work of their name-alike, with optional locking via extra arguments, and are called by their name-alike. The goal was to have the original functions so they would behave exactly as before. Both PJ and I tested these fixes, and the deadlocking problem is no longer encountered. (closes issue #14357) Reported by: pj Patches: 14357.diff uploaded by murf (license 17) Tested by: pj, murf ........ 2009-02-18 06:15 +0000 [r176903-176906] Russell Bryant * include/asterisk/heap.h, /: Merged revisions 176904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009) | 2 lines Add example code for a heap traversal. ........ * main/pbx.c, /: Merged revisions 176901 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 | russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines Fix a number of incorrect uses of strncpy(). The big problem here is that the 3rd argument provided in these uses of strncpy() did not reserve a byte for the null terminator, leaving the potential for writing one byte past the end of the buffer. Aside from this, there were coding guidelines violations with regards to spacing, as well as hard coded lengths being used instead of sizeof(). ........ 2009-02-18 00:23 +0000 [r176809] Shaun Ruffell * /, codecs/codec_dahdi.c: Merged revisions 176760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. ........ 2009-02-17 22:21 +0000 [r176731] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 176705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. ........ 2009-02-17 22:15 +0000 [r176711] Jeff Peeler * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 176708 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ 2009-02-17 21:41 +0000 [r176699] Mark Michelson * include/asterisk/frame.h, /: Merged revisions 176697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in frame.h ........ 2009-02-17 21:24 +0000 [r176675] Russell Bryant * main/timing.c, main/channel.c, /, res/res_timing_pthread.c, res/res_timing_dahdi.c, include/asterisk/timing.h: Merged revisions 176666 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ 2009-02-17 21:16 +0000 [r176644] Tilghman Lesher * res/res_odbc.c, channels/chan_local.c, /: Merged revisions 176592,176642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176592 | tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines Add assertions in the quest to track down a refcount leak. (closes issue #14485) Reported by: davevg ........ r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ 2009-02-17 20:57 +0000 [r176559-176637] Russell Bryant * tests/test_heap.c (added), /: Merged revisions 176635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17 Feb 2009) | 4 lines Add a test module for the heap implementation. Review: http://reviewboard.digium.com/r/160/ ........ * include/asterisk/heap.h (added), /, main/Makefile, main/heap.c (added): Merged revisions 176632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ ........ * apps/app_queue.c, main/pbx.c, /: Merged revisions 176557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 Feb 2009) | 12 lines Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis ........ 2009-02-17 14:48 +0000 [r176461-176503] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 176501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176501 | tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines In this version, we can combine the queries, because we support dropping nonexistent columns. ........ * /, channels/chan_sip.c: Merged revisions 176459 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines Merged revisions 176426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ ................ 2009-02-16 23:57 +0000 [r176362] David Vossel * channels/chan_iax2.c, /: Merged revisions 176355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ ................ 2009-02-16 23:17 +0000 [r176321] Tilghman Lesher * /, channels/chan_skinny.c: Merged revisions 176320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009) | 7 lines Use the correct list macros for deleting an item from the middle of a list. (issue #13777) Reported by: pj Patches: 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14) Tested by: pj ........ 2009-02-16 22:00 +0000 [r176259] Kevin P. Fleming * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 176255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ 2009-02-16 21:50 +0000 [r176257] Mark Michelson * /, apps/app_meetme.c: Merged revisions 176253 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ ................ 2009-02-16 21:36 +0000 [r176251] David Vossel * channels/chan_iax2.c, /: Merged revisions 176248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ ................ 2009-02-16 18:38 +0000 [r176176] Mark Michelson * /, main/logger.c: Merged revisions 176174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. ........ 2009-02-16 17:10 +0000 [r176102] Russell Bryant * /, channels/chan_features.c (removed): Merged revisions 176100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 Feb 2009) | 4 lines Remove chan_features. Review: http://reviewboard.digium.com/r/161/ ........ 2009-02-16 17:07 +0000 [r176099] Tilghman Lesher * configs/func_odbc.conf.sample: Eliminate mention of a variable which exists only in trunk. (Thanks, jsmith) 2009-02-16 15:38 +0000 [r176032] Joshua Colp * /, channels/chan_sip.c: Merged revisions 176030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines Merged revisions 176029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ ................ 2009-02-16 09:42 +0000 [r176023] Michiel van Baak * include/asterisk/manager.h, doc/unistim.txt, channels/chan_unistim.c, /, channels/chan_sip.c: Merged revisions 175952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ 2009-02-15 21:28 +0000 [r175831-175890] Russell Bryant * main/sched.c, /, include/asterisk/sched.h: Merged revisions 175882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines Make ast_sched_report() and ast_sched_dump() thread safe. ........ * main/sched.c, /, channels/chan_sip.c, include/asterisk/sched.h: Merged revisions 175829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. ........ 2009-02-13 20:48 +0000 [r175662] David Vossel * channels/chan_iax2.c, configs/iax.conf.sample, channels/iax2.h: Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ 2009-02-13 19:52 +0000 [r175593] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 175591 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600 (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ ................ 2009-02-13 16:44 +0000 [r175551] Joshua Colp * /, apps/app_record.c: Merged revisions 175549 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ 2009-02-12 21:41 +0000 [r175370] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ 2009-02-12 21:27 +0000 [r175342] Tilghman Lesher * main/udptl.c, /: Merged revisions 175334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ 2009-02-12 20:51 +0000 [r175300] Jeff Peeler * /, main/features.c: Merged revisions 175298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ 2009-02-12 20:48 +0000 [r175257-175297] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ 2009-02-12 18:50 +0000 [r175251] Kevin P. Fleming * channels/chan_iax2.c, /: Merged revisions 175250 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line correct warning message to not refer specifically to DAHDI ........ 2009-02-12 18:01 +0000 [r175190] Jeff Peeler * /, main/features.c: Merged revisions 175188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ 2009-02-12 17:09 +0000 [r175130] David Vossel * channels/chan_iax2.c, /: Merged revisions 175127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009) | 4 lines Setting key rotation to be off by default Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. ........ 2009-02-12 17:08 +0000 [r175129] Russell Bryant * main/rtp.c, /: Merged revisions 175125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-12 16:35 +0000 [r174947-175123] Mark Michelson * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 175121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ * apps/app_queue.c, /: Merged revisions 174951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174951 | mmichelson | 2009-02-11 17:12:57 -0600 (Wed, 11 Feb 2009) | 3 lines Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic ........ * apps/app_queue.c, /: Merged revisions 174948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 20 lines Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy ........ * apps/app_dial.c, main/channel.c, main/pbx.c, /, apps/app_dictate.c, apps/app_waitforsilence.c, include/asterisk/channel.h: Merged revisions 174945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ 2009-02-11 14:46 +0000 [r174846] Joshua Colp * main/channel.c, /: Merged revisions 174844 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174844 | file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ 2009-02-10 23:21 +0000 [r174769-174823] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 lines Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa ........ * main/manager.c, /: Merged revisions 174764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 lines Fix an fd leak that would occur in HTTP AMI sessions The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ ........ 2009-02-10 20:17 +0000 [r174714] Joshua Colp * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines Only decrease inringing count if above zero. (issue #13238) Reported by: kowalma ........ 2009-02-10 18:18 +0000 [r174590] Matthew Nicholson * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb 2009) | 25 lines Merged revisions 174583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ ................ 2009-02-10 17:49 +0000 [r174545-174582] Joshua Colp * /, channels/chan_sip.c: Merged revisions 174580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174580 | file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines Set the type for the peer structure to be a peer as the default. (closes issue #14447) Reported by: triccyx ........ * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma ........ 2009-02-10 07:07 +0000 [r174471-174504] Tilghman Lesher * apps/app_stack.c, apps/app_voicemail.c, /: Merged revisions 174503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174503 | tilghman | 2009-02-10 01:06:29 -0600 (Tue, 10 Feb 2009) | 2 lines Fix0ring build ........ * apps/app_stack.c, /: Merged revisions 174470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174470 | tilghman | 2009-02-09 23:39:33 -0600 (Mon, 09 Feb 2009) | 2 lines Remove the usage of the KeepAlive app, as it no longer exists. ........ 2009-02-10 05:13 +0000 [r174428-174440] Steve Murphy * apps/app_osplookup.c: This patch corrects warnings which seem to appear only on 64-bit compilers, gcc-4.3.2. * apps/app_rpt.c: One final fix in the 1.6.1 release only; some variables the compiler worries "may not be initialized". * apps/app_rpt.c, /: Merged revisions 174435 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174435 | murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c. (closes issue #14435) Reported by: D_McNaul ........ * apps/app_rpt.c, /: Merged revisions 174432 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174432 | murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines More intptr_t work. ........ * apps/app_rpt.c, /: Merged revisions 174370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) | 10 lines Merged revisions 174369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ ................ 2009-02-09 17:47 +0000 [r174330] David Vossel * /, apps/app_externalivr.c: Merged revisions 174325 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174325 | dvossel | 2009-02-09 11:26:02 -0600 (Mon, 09 Feb 2009) | 9 lines Fixes issue with hangups not being sent and external process never terminating. The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. (closes issue #14251) Reported by: chris-mac Tested by: dvossel ........ 2009-02-09 17:30 +0000 [r174326-174329] Mark Michelson * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines Fix something I messed up in the merge I just did ........ * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines Merged revisions 174282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ ................ 2009-02-09 14:50 +0000 [r174221] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn ........ ................ 2009-02-07 16:18 +0000 [r174154] Russell Bryant * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) | 10 lines Merged revisions 174148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ ................ 2009-02-07 00:09 +0000 [r174086] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines Merged revisions 174082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ ................ 2009-02-06 19:30 +0000 [r173994-174043] Joshua Colp * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription. (closes issue #14322) Reported by: amessina ........ * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines Merged revisions 173967-173968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ ................ 2009-02-06 17:05 +0000 [r173964-173966] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h: revert revision 173964 * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h: Merged revisions 173952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ 2009-02-06 16:01 +0000 [r173904] Joshua Colp * apps/app_chanspy.c, /, main/audiohook.c: Merged revisions 173902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb 2009) | 4 lines Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. (closes issue #14414) Reported by: bluecrow76 ........ 2009-02-06 10:26 +0000 [r173850] Russell Bryant * main/manager.c, /: Merged revisions 173848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173848 | russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines Resolve a memory leak that would occur on an invalid channel given to Action: Status ........ 2009-02-05 23:53 +0000 [r173779] Mark Michelson * configs/extensions.conf.sample, /: Merged revisions 173776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up ........ 2009-02-05 23:51 +0000 [r173778] Tilghman Lesher * res/res_config_sqlite.c: Oops, merge from trunk broke 1.6.1 2009-02-05 23:31 +0000 [r173775] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) ........ 2009-02-05 21:06 +0000 [r173699] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ ................ 2009-02-05 20:35 +0000 [r173695] Mark Michelson * apps/app_queue.c, /: Merged revisions 173693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ ................ 2009-02-05 19:37 +0000 [r173658] Tilghman Lesher * res/res_config_sqlite.c, /: Merged revisions 173657 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173657 | tilghman | 2009-02-05 13:36:29 -0600 (Thu, 05 Feb 2009) | 2 lines Change the first field, or we don't get the necessary field separation. ........ 2009-02-05 18:50 +0000 [r173541-173595] Mark Michelson * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ ................ * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ ................ * apps/app_queue.c, /: Merged revisions 173507 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r ........ 2009-02-04 21:32 +0000 [r173506] David Vossel * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions 173502 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173502 | dvossel | 2009-02-04 15:25:14 -0600 (Wed, 04 Feb 2009) | 9 lines Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884 Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel ........ 2009-02-04 21:28 +0000 [r173505] Jeff Peeler * include/asterisk/features.h, /, main/features.c: Merged revisions 173500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ ................ 2009-02-04 18:52 +0000 [r173459] Tilghman Lesher * main/tcptls.c, /: Merged revisions 173458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) ........ 2009-02-04 17:46 +0000 [r173356-173399] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ ................ * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ ................ * /, main/file.c: Merged revisions 173354 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut ........ 2009-02-04 00:46 +0000 [r173313] Tilghman Lesher * main/pbx.c, /: Merged revisions 173311 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ 2009-02-03 00:26 +0000 [r173115] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 173104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ ................ 2009-02-03 00:01 +0000 [r173069] Terry Wilson * /, main/features.c: Merged revisions 173067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ ................ 2009-02-02 18:15 +0000 [r172895] Leif Madsen * /, configs/res_ldap.conf.sample: Merged revisions 172894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert ........ 2009-02-01 02:45 +0000 [r172708-172743] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen ........ * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) | 7 lines Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) ........ 2009-01-31 00:07 +0000 [r172636-172638] Terry Wilson * configs/features.conf.sample, /: Merged revisions 172581 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 Jan 2009) | 2 lines Remove incorret line from sample config ........ * CHANGES, configs/features.conf.sample, apps/app_dial.c, main/global_datastores.c, /, main/features.c, include/asterisk/global_datastores.h: Merged revisions 172580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ 2009-01-30 22:24 +0000 [r172609] Mark Michelson * /, include/asterisk/channel.h: Merged revisions 172598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h ........ 2009-01-30 08:27 +0000 [r172509] Olle Johansson * CHANGES: Remove an extra "the" and restructure a bit 2009-01-29 23:53 +0000 [r172504] Tilghman Lesher * apps/app_rpt.c, main/asterisk.c, /, autoconf/ast_func_fork.m4, configure, main/app.c: Merged revisions 172441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ 2009-01-29 22:05 +0000 [r172435] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 172400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. ........ 2009-01-29 20:54 +0000 [r172317-172402] Tilghman Lesher * utils/muted.c, /: Merged revisions 146514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk (closes issue #14360) Reported by: oej ........ r146514 | russell | 2008-10-05 17:11:30 -0500 (Sun, 05 Oct 2008) | 2 lines Make this build on my mac. ........ * configs/func_odbc.conf.sample, /: Merged revisions 172315 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 Jan 2009) | 2 lines Better document mode=multirow, based upon a conversation with Jared. ........ 2009-01-29 13:50 +0000 [r172272] Leif Madsen * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a couple of fields. closes issue #14339) Reported by: fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) ........ 2009-01-29 11:24 +0000 [r172218-172235] Olle Johansson * /, channels/chan_sip.c: Merged revisions 172234 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172234 | oej | 2009-01-29 12:19:29 +0100 (Tor, 29 Jan 2009) | 7 lines Make sure register= line supports both port and expiry at the same time. (closes issue #14185) Reported by: Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) Tested by: Nick_Lewis ........ * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ ................ * /, configs/sip.conf.sample: Merged revisions 171880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 lines Add some more notes about device matching. ........ 2009-01-28 Leif Madsen * Asterisk 1.6.1-rc1 released 2009-01-28 22:52 +0000 [r172133] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 172131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172131 | tilghman | 2009-01-28 16:48:01 -0600 (Wed, 28 Jan 2009) | 7 lines Fix how we skip fields (to avoid fields which don't exist) when doing an UPDATE. (closes issue #14205) Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ 2009-01-28 20:56 +0000 [r172067] Steve Murphy * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 172063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ 2009-01-28 17:29 +0000 [r171966] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ ................ 2009-01-28 13:21 +0000 [r171857] Olle Johansson * configs/sip.conf.sample: Merged revisions 171838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ 2009-01-27 22:01 +0000 [r171620-171693] Mark Michelson * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ ................ * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan 2009) | 26 lines Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ ................ * apps/app_queue.c, /: Merged revisions 171618 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario ........ 2009-01-27 15:19 +0000 [r171540] Olle Johansson * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 lines Solving the same issue, but a bit different in trunk... Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ ................ 2009-01-26 14:58 +0000 [r171361] Olle Johansson * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | 17 lines Merged revisions 171264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-01-26 00:04 +0000 [r171190] Tilghman Lesher * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) | 13 lines Merged revisions 171187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2009-01-25 13:40 +0000 [r170982] Sean Bright * /, apps/app_page.c: Merged revisions 170980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ ................ 2009-01-25 02:52 +0000 [r170945] Russell Bryant * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. ........ 2009-01-24 13:57 +0000 [r170839] Tilghman Lesher * configs/res_odbc.conf.sample, /: Merged revisions 170837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ ................ 2009-01-23 23:53 +0000 [r170831] Richard Mudgett * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line Fix asterisk.pdf generation if branch name has an underscore in it. ........ 2009-01-23 22:59 +0000 [r170792] Russell Bryant * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 | russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines Don't blow up if a branch name has an underscore in it ........ 2009-01-23 20:57 +0000 [r170693-170722] Mark Michelson * configs/res_odbc.conf.sample, /: Merged revisions 170720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines Add notes to the idlecheck explanation in res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) ........ ................ * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600 (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use deprecated syntax * Convert Wait,1 to Wait(1) * Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all priorities beyond the first Also added test for Chinese numbers, too. (closes issue #14320) Reported by: dant Patches: i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license 670) ........ ................ 2009-01-23 20:20 +0000 [r170664] Joshua Colp * main/channel.c, /: Merged revisions 170652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ ................ 2009-01-23 19:37 +0000 [r170637] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 170608 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170608 | tilghman | 2009-01-23 13:25:10 -0600 (Fri, 23 Jan 2009) | 9 lines Merged revisions 170588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 Jan 2009) | 2 lines Additions to AST-2009-001 ........ ................ 2009-01-23 19:10 +0000 [r170507-170571] Joshua Colp * apps/app_dial.c, /: Merged revisions 170569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ ................ * /, channels/chan_sip.c: Merged revisions 170505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold. (closes issue #14295) Reported by: klaus3000 ........ ................ 2009-01-23 17:49 +0000 [r170502] Michiel van Baak * /, channels/chan_h323.c: Merged revisions 170501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170501 | mvanbaak | 2009-01-23 18:46:02 +0100 (Fri, 23 Jan 2009) | 1 line let's use SENTINEL where needed ........ 2009-01-23 16:35 +0000 [r170458] Doug Bailey * channels/chan_dahdi.c: MWI messages included in CID spill was not being properly handled and prevented the call from being processed (issue #14313) Reported by: seandarcy Tested by: dbailey 2009-01-23 15:51 +0000 [r170395] Mark Michelson * main/channel.c, /: Merged revisions 170393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan 2009) | 36 lines Merged revisions 170392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut ........ ................ 2009-01-22 20:06 +0000 [r170242] Joshua Colp * main/rtp.c, /: Merged revisions 170240 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) | 14 lines Merged revisions 170239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines Don't crash if RTCP is not enabled on an RTP structure but statistics are output. (closes issue #14234) Reported by: jcovert Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ ................ 2009-01-22 17:21 +0000 [r170178] Tilghman Lesher * pbx/pbx_config.c, /: Merged revisions 170165 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170165 | tilghman | 2009-01-22 11:19:28 -0600 (Thu, 22 Jan 2009) | 13 lines Merged revisions 170158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009) | 6 lines Allow global variables after substitution to be as long as other variables. (closes issue #14263) Reported by: markd Patches: 20090120__bug14263.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2009-01-22 16:54 +0000 [r170049-170150] Joshua Colp * /, apps/app_meetme.c: Merged revisions 170148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines Merged revisions 170147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ ................ * main/pbx.c, /: Merged revisions 170051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) | 13 lines Merged revisions 170050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness. (closes issue #14011) Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga (license 665) ........ ................ * apps/app_parkandannounce.c, /: Merged revisions 170047 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop. (closes issue #14304) Reported by: jcovert ........ 2009-01-22 00:46 +0000 [r169946] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 169944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169944 | tilghman | 2009-01-21 18:44:52 -0600 (Wed, 21 Jan 2009) | 16 lines Merged revisions 169943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 ........ ................ 2009-01-21 23:28 +0000 [r169871] Joshua Colp * main/pbx.c, /: Merged revisions 169869 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) | 11 lines Merged revisions 169867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 lines Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed. (closes issue #13839) Reported by: mcallist ........ ................ 2009-01-21 22:23 +0000 [r169830] Michiel van Baak * /, doc/tex/extensions.tex: Merged revisions 169793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009) | 2 lines remove duplicated sentence. ........ 2009-01-21 22:11 +0000 [r169792-169796] Mark Michelson * /, main/say.c: Merged revisions 169794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169794 | mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17 lines Fix a crash when saying certain numbers in Chinese This commit fixes a crash that was occurring when attempting to say a number between 10000 and 100000 due to dividing by 0. This also removes some places where a "zero" is spoken when it should not be. (closes issue #14291) Reported by: dant Patches: say.c-14291.diff uploaded by dant (license 670) Tested by: dant ........ * /, channels/chan_sip.c: Merged revisions 169791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169791 | mmichelson | 2009-01-21 15:53:55 -0600 (Wed, 21 Jan 2009) | 18 lines Further fix some oddities in sip show users and sip show peers logic ccesario on IRC pointed out that his sip peers were not displayed properly when he would issue the command "sip show peers." The problem was that the onlymatchonip field was used to determine if the endpoint was a "peer" or "user." The tricky part is that a "friend" is supposed to be treated as both a "user" and a "peer" but the logic would not allow "friends" to show up as "peers" since onlymatchonip was set to FALSE for friends. I have modified the sip_peer structure to more explicitly keep track of what type endpoint it is so that the various manager and CLI commands will display the expected information Reported by ccesario via IRC Tested by ccesario ........ 2009-01-21 21:05 +0000 [r169725] Tilghman Lesher * main/asterisk.c, /: Merged revisions 169723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009) | 15 lines Merged revisions 169722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines Extra NULLs in the output cause some terminal types to abort in the middle of a color code, causing terminal weirdness. (closes issue #14130) Reported by: coolmig Patches: 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, coolmig ........ ................ 2009-01-21 17:40 +0000 [r169674] Steve Murphy * utils/refcounter.c, /: Merged revisions 169673 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169673 | murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines This patch corrects a segfault reported in 14289, due to a null ptr being refd. Yes, seanbright is right in the bug comments, that is the fix. Sorry for this oversight; I guess my personal usage didn't have this happen! murf (closes issue #14289) Reported by: jamesgolovich ........ 2009-01-21 10:49 +0000 [r169622-169626] Russell Bryant * /: Merged revisions 169625 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169625 | russell | 2009-01-21 04:49:00 -0600 (Wed, 21 Jan 2009) | 2 lines Remove properties that erroneously got merged into trunk ........ * main/tcptls.c, /: Merged revisions 169620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169620 | russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines Fix a regression in TCP support. This patch fixes a problem that caused chan_sip to think that every open TCP session was to a remote address of 0.0.0.0:0. (closes issue #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt uploaded by jamesgolovich (license 176) ........ 2009-01-21 00:35 +0000 [r169559-169613] Mark Michelson * apps/app_queue.c, /: Merged revisions 169611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines Fix device state parsing issues for channel names with multiple slashes The fix being applied is a bit different for trunk and the 1.6.X branches. For trunk, we only wish to strip off the characters beyond the second slash if the channel is a Local channel (i.e. we are removing the /n from the device name). Other channel technologies with multiple slashes (e.g. DAHDI) need the information after the second slash in order to get the proper device state information. In addition to this fix, the 1.6.X branches are receiving a much more important fix as well. The problem in 1.6.X is that the member's device name was being directly changed instead of having a copy changed. This meant that we would strip off the second slash and trailing characters and then leave the member's device name like that permanently thereafter. (closes issue #14014) Reported by: kebl0155 Patches: 14014_number2.patch uploaded by putnopvut (license 60) Tested by: kebl0155 ........ * apps/app_queue.c, /: Merged revisions 169574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169574 | mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6 lines Use the default timeout for a queue instead of -1 (closes issue #14272) Reported by: timking ........ * /, channels/chan_sip.c: Merged revisions 169557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169557 | mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19 lines Convert the character pointers in a sip_request to be pointer offsets When an ast_str expands to hold more data, any pointers that were pointing to the data prior to the expansion will be pointing at invalid memory. This change makes such pointers used in chan_sip.c instead be offsets from the beginning of the string so that the same math may be applied no matter where in memory the string resides. To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so that given a sip_request and an offset, the string at that offset is returned. (closes issue #14220) Reported by: riksta Tested by: putnopvut Review http://reviewboard.digium.com/r/126/ ........ 2009-01-20 19:31 +0000 [r169488-169554] Terry Wilson * /, main/features.c: Merged revisions 169510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169510 | twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines Make a proper builtin attended transfer to parking work This is an ugly hack from 1.4 that allows the timeout callback from a parked call to use the right channel name for the callback when the park is done with a builtin attended transfer (that isn't completed early). This hasn't ever worked in trunk and no one has complained yet, so eh. ........ * /, main/features.c: Merged revisions 169486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009) | 13 lines Merged revisions 169485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) | 6 lines Don't play audio to the channel if we've masqueraded (closes issue #14066) Reported by: bluefox Tested by: otherwiseguy, bluefox ........ ................ 2009-01-19 20:10 +0000 [r169368] Tilghman Lesher * main/manager.c, /, apps/app_userevent.c: Merged revisions 169365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169365 | tilghman | 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) | 11 lines Merged revisions 169364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian ........ ................ 2009-01-19 15:55 +0000 [r169213] Mark Michelson * channels/chan_local.c, /: Merged revisions 169211 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169211 | mmichelson | 2009-01-19 09:54:06 -0600 (Mon, 19 Jan 2009) | 21 lines Merged revisions 169210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a potential NULL pointer dereference Move the check for if both channels on a local_pvt have generators to below where p->chan is checked for NULLity (NULLness?). This prevents a crash from occurring if p->chan is NULL. (closes issue #14189) Reported by: sascha Patches: 14189.patch uploaded by putnopvut (license 60) Tested by: sascha ........ ................ 2009-01-17 18:46 +0000 [r169154] Doug Bailey * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add discriminator for when ring pulse alert signal is used to preface MWI spills This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up 2009-01-17 01:59 +0000 [r168981-169082] Terry Wilson * main/tcptls.c, /, main/http.c, include/asterisk/tcptls.h: Merged revisions 169080 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169080 | twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines Fix qualify for TCP peer (closes issue #14192) Reported by: pabelanger Patches: asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich ........ * /, channels/chan_sip.c: Merged revisions 169044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169044 | twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines Fix port :0 added to SIP INVITE URI when outboundproxy used (closes issue #14233) Reported by: chris-mac Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy ........ * /, main/features.c: Merged revisions 168941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009) | 19 lines Merged revisions 168716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines Convert call to park_call_full to masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded parking, otherwise we will try to call ast_hangup() in __pbx_run() and in do_parking_thread() and then promptly crash. (closes issue #14215) Reported by: waverly360 Tested by: otherwiseguy (closes issue #14228) Reported by: kobaz Tested by: otherwiseguy ........ ................ 2009-01-16 22:46 +0000 [r168979] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168976 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan 2009) | 26 lines Merged revisions 168975 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines Account for possible NULL pointer when we receive a 408 in response to a REGISTER It may be that by the time we receive a reply to a REGISTER request, the attempt has timed out and thus the registry structure pointed to by the corresponding sip_pvt has gone away. This situation was handled properly for a 200 OK response, but the 408 case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash This commit fixes this assumption and prints out a message to the console if we should receive a late 408 response to a REGISTER (closes issue #14211) Reported by: aborghi Patches: 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi ........ ................ 2009-01-16 18:55 +0000 [r168836] Tilghman Lesher * include/asterisk/say.h, apps/app_voicemail.c, /, main/say.c: Merged revisions 168832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ ................ 2009-01-16 00:47 +0000 [r168739-168748] Steve Murphy * res/ael/pval.c, /: Merged revisions 168746 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | 20 lines Merged revisions 168745 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines This patch fixes a problem where a goto (or jump, in this case) fails a consistency check because it can't find a matching extension. The problem was a missing instruction to end the range notation in the code where it converts the pattern into a regex and uses the regex code to determine the match. I tested using the AEL code the user supplied, and now, the consistency check passes. (closes issue #14141) Reported by: dimas ........ ................ * main/ast_expr2.c, /, main/ast_expr2.h, main/ast_expr2.y: Merged revisions 168737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168737 | murf | 2009-01-15 13:54:59 -0700 (Thu, 15 Jan 2009) | 16 lines This patch allows null args in ast_expr2 func calls, and fixes commas being converted to pipes, which was 1.4 type stuff. If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it won't complain about the empty arg (c,,...) and fabled's patch won't let it swap the commas for pipes. Ran it thru my dialplan and no complaints. (closes issue #14169) Reported by: fabled Patches: function-argument-separator-fix.diff uploaded by fabled (license 448) ........ 2009-01-15 19:17 +0000 [r168729] Mark Michelson * channels/chan_sip.c: Merged revisions 168728 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168728 | mmichelson | 2009-01-15 13:16:29 -0600 (Thu, 15 Jan 2009) | 3 lines Fix the compactheaders option in sip.conf ........ 2009-01-15 19:05 +0000 [r168727] Olle Johansson * /, configs/extconfig.conf.sample: Merged revisions 168722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168722 | oej | 2009-01-15 19:47:14 +0100 (Tor, 15 Jan 2009) | 10 lines Merged revisions 168721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines Meetme actually has realtime but wasn't documented ........ ................ 2009-01-15 19:00 +0000 [r168726] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168725 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168725 | mmichelson | 2009-01-15 13:00:06 -0600 (Thu, 15 Jan 2009) | 17 lines Remove an unneeded condition for line addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically allocated buffer to hold the text of the request. There was a check in the add_line function to not attempt to write the line into the buffer if we did not have room for it. In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str structure is used to hold the text. Since it may grow to fit an arbitrarily sized string, this check in add_line is no longer valid. I found this oddity while attempting to fix issue #14220; however, I do not believe that this is the fix for that issue since the output supplied by the reporter did not contain the warning message that would be printed had this condition been satisfied. ........ 2009-01-15 18:20 +0000 [r168714-168715] Olle Johansson * /, configs/sip.conf.sample: Merged revisions 168711 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. ........ * /, channels/chan_sip.c: Merged revisions 168712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168712 | oej | 2009-01-15 19:08:59 +0100 (Tor, 15 Jan 2009) | 3 lines Make sure that we have the same terminology in sip.conf.sample and the source code warning. Thanks Nick Lewis for pointing this out in the bug tracker. ........ 2009-01-15 15:37 +0000 [r168707] Sean Bright * /, apps/app_meetme.c: Merged revisions 168705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines Add a missing unlock and properly handle the 'maxusers' setting on MeetMe conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop ........ 2009-01-15 00:15 +0000 [r168631] Mark Michelson * apps/app_queue.c, /: Merged revisions 168629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines Merged revisions 168628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines Fix some crashes from bad datastore handling in app_queue.c * The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr ........ ................ 2009-01-14 21:55 +0000 [r168625] Richard Mudgett * channels/misdn/isdn_lib.c, /: Merged revisions 168623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168623 | rmudgett | 2009-01-14 15:51:06 -0600 (Wed, 14 Jan 2009) | 11 lines Merged revisions 168622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) | 4 lines * Fixed create_process() allocation of process ID values. The allocated process IDs could overflow their respective NT and TE fields. Affects outgoing calls. ........ ................ 2009-01-14 21:30 +0000 [r168621] Steve Murphy * /, apps/app_page.c: Merged revisions 168613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9 lines Merged revisions 168608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ ................ 2009-01-14 21:00 +0000 [r168618] Sean Bright * contrib/scripts/autosupport, /: Merged revisions 168615 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168615 | seanbright | 2009-01-14 15:58:26 -0500 (Wed, 14 Jan 2009) | 16 lines Merged revisions 168614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan 2009) | 9 lines Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of zttest. (closes issue #14132) Reported by: dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded by dsedivec (license 638) ........ ................ 2009-01-14 20:18 +0000 [r168611] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168610 | mmichelson | 2009-01-14 14:13:48 -0600 (Wed, 14 Jan 2009) | 9 lines Restore the "sip show users" and "sip show user" CLI commands (closes issue #14180) Reported by: amorsen Patches: sip_show_users_161v3.diff uploaded by putnopvut (license 60) Tested by: blitzrage, amorsen ........ 2009-01-14 19:12 +0000 [r168606] Tilghman Lesher * main/udptl.c, /: Merged revisions 168604 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168604 | tilghman | 2009-01-14 13:11:14 -0600 (Wed, 14 Jan 2009) | 14 lines Merged revisions 168603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines Don't read into a buffer without first checking if a value is beyond the end. (closes issue #13600) Reported by: atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 (license 14) Tested by: atis ........ ................ 2009-01-14 02:11 +0000 [r168582-168596] Terry Wilson * /, apps/app_page.c: Merged revisions 168594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines Merged revisions 168593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ ................ * /, res/res_http_post.c: Merged revisions 168588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168588 | twilson | 2009-01-13 17:05:43 -0600 (Tue, 13 Jan 2009) | 5 lines Fully overwrite a same-named file when uploading (closes issue #14190) Reported by: timking ........ * /, channels/chan_sip.c: Merged revisions 168578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168578 | twilson | 2009-01-13 16:22:34 -0600 (Tue, 13 Jan 2009) | 14 lines Merged revisions 168551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines Don't pass a value with a side effect to a macro (closes issue #14176) Reported by: paraeco Patches: chan_sip.c.diff uploaded by paraeco (license 658) ........ ................ 2009-01-13 19:35 +0000 [r168565] Russell Bryant * main/indications.c, main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_channel.c, main/app.c, res/snmp/agent.c, res/res_indications.c, channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /, include/asterisk/indications.h, apps/app_readexten.c, apps/app_disa.c, include/asterisk/channel.h: Merged revisions 168562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ 2009-01-13 17:52 +0000 [r168528-168549] Tilghman Lesher * /, funcs/func_logic.c: Merged revisions 168547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168547 | tilghman | 2009-01-13 11:51:12 -0600 (Tue, 13 Jan 2009) | 13 lines Merged revisions 168546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) | 6 lines If either conditional is NULL, don't try copying it. (closes issue #14226) Reported by: caspy Patches: 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, channels/chan_alsa.c: Merged revisions 168526 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168526 | tilghman | 2009-01-12 17:45:51 -0600 (Mon, 12 Jan 2009) | 12 lines Merged revisions 167095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines Repeat attempts to write when we receive -EAGAIN from the driver, as detailed in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) Reported by: Jerry Geis (via the -users list) Fixed by: me (license 14) ........ ................ 2009-01-12 23:13 +0000 [r168524] Mark Michelson * main/srv.c, /: Merged revisions 168523 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168523 | mmichelson | 2009-01-12 17:12:30 -0600 (Mon, 12 Jan 2009) | 11 lines bump the verbosity of a message in srv.c up by one. It used to be at this level prior to a large patch merge which converted ast_verbose calls to ast_verb (closes issue #14221) Reported by: jcovert Patches: srv.c.patch uploaded by jcovert (license 551) ........ 2009-01-12 22:00 +0000 [r168510-168519] Jeff Peeler * /, res/res_agi.c: Merged revisions 168517 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168517 | jpeeler | 2009-01-12 15:51:46 -0600 (Mon, 12 Jan 2009) | 12 lines Merged revisions 168516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines (closes issue #13881) Reported by: hoowa Update the app CDR field for AGI commands that are not executing an application via "exec". ........ ................ * /, channels/chan_agent.c: Merged revisions 168508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168508 | jpeeler | 2009-01-12 14:53:04 -0600 (Mon, 12 Jan 2009) | 15 lines Merged revisions 168507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG Tested by: denisgalvao This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock. Review: http://reviewboard.digium.com/r/35/ ........ ................ 2009-01-12 17:26 +0000 [r168500] Olle Johansson * /, apps/app_minivm.c: Merged revisions 168497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168497 | oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines Better to use the proper app name ........ 2009-01-12 15:05 +0000 [r168488] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ 2009-01-12 14:58 +0000 [r168484] Russell Bryant * /, configs/indications.conf.sample: Merged revisions 168481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168481 | russell | 2009-01-12 08:57:49 -0600 (Mon, 12 Jan 2009) | 10 lines Merged revisions 168480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ ................ 2009-01-10 01:44 +0000 [r168336] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 168334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168334 | tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines sizeof for a stringfield is 4. Kinda low for reconstructing a field value. ........ 2009-01-09 23:18 +0000 [r168272] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 168270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan 2009) | 9 lines Merged revisions 168267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line update to use new sound file packages that include license files ........ ................ 2009-01-09 23:12 +0000 [r168266] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 168192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168192 | rmudgett | 2009-01-09 15:43:30 -0600 (Fri, 09 Jan 2009) | 10 lines Merged revisions 168191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 * Miscellaneous doxygen comments added. ........ ................ 2009-01-09 22:23 +0000 [r168209] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 168200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168200 | russell | 2009-01-09 16:21:05 -0600 (Fri, 09 Jan 2009) | 10 lines Merged revisions 168198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines Make this compile for mvanbaak ........ ................ 2009-01-09 21:57 +0000 [r168196] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168193 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan 2009) | 21 lines Merged revisions 168128 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines Add check_via calls to more request handlers INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not checking the topmost Via to determine where to send the response. Adding check_via calls to those request handlers solves this. (closes issue #13071) Reported by: baron Patches: check_via.patch uploaded by baron (license 531) Tested by: baron ........ ................ 2009-01-09 20:30 +0000 [r168157] Terry Wilson * /, res/res_phoneprov.c: Merged revisions 168142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168142 | twilson | 2009-01-09 14:25:25 -0600 (Fri, 09 Jan 2009) | 7 lines Don't leak memory if phoneprov.conf does not exist (closes issue #14203) Reported by: jamesgolovich Patches: asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich (license 176) ........ 2009-01-09 18:42 +0000 [r168092] Tilghman Lesher * /, res/res_agi.c: Merged revisions 168090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168090 | tilghman | 2009-01-09 12:30:55 -0600 (Fri, 09 Jan 2009) | 3 lines When using ast_str with a non-ast_str-enabled API, we need to update the buffer or otherwise, we cannot use ast_str_strlen(). ........ 2009-01-09 16:41 +0000 [r168015] Matthew Nicholson * /, main/logger.c: Merged revisions 168014 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168014 | mnicholson | 2009-01-09 10:32:34 -0600 (Fri, 09 Jan 2009) | 5 lines Use ast_safe_system() in logger.c instead of system() (closes issue #14194) Reported by: pabelanger ........ 2009-01-09 00:45 +0000 [r167972] Terry Wilson * apps/app_dial.c, /: Merged revisions 167935 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set ........ 2009-01-08 22:45 +0000 [r167836-167905] Tilghman Lesher * /, res/res_agi.c: Merged revisions 167894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167894 | tilghman | 2009-01-08 16:37:20 -0600 (Thu, 08 Jan 2009) | 13 lines Merged revisions 167840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines Don't truncate database results at 255 chars. (closes issue #14069) Reported by: evandro Patches: 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, apps/app_minivm.c: Merged revisions 167835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167835 | tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines Textual changes, consistency in status variable naming, and other minor bugs. (closes issue #13943) Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded by Marquis (license 32) ........ 2009-01-08 17:28 +0000 [r167701-167727] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 167720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan 2009) | 9 lines Merged revisions 167714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line remove an unnecessary argument to queue_request() ........ ................ * /, channels/chan_sip.c: Merged revisions 167700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan 2009) | 12 lines Merged revisions 167620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available. http://reviewboard.digium.com/r/123/ ........ ................ 2009-01-08 14:30 +0000 [r167663] Leif Madsen * contrib/scripts/sip-friends.sql, /: Merged revisions 167662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167662 | lmadsen | 2009-01-08 09:27:53 -0500 (Thu, 08 Jan 2009) | 1 line Oops... fix the fieldname I changed yesterday to be right. ........ 2009-01-07 22:37 +0000 [r167544-167573] Russell Bryant * /, main/file.c: Merged revisions 167569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167569 | russell | 2009-01-07 16:36:34 -0600 (Wed, 07 Jan 2009) | 10 lines Merged revisions 167566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines Fix the last couple of places where free() was improperly used directly. ........ ................ * /, main/file.c: Merged revisions 167555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167555 | russell | 2009-01-07 16:27:23 -0600 (Wed, 07 Jan 2009) | 10 lines Merged revisions 167554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines Don't fclose() the file early, the filestream destructor will handle it. ........ ................ * /, main/file.c: Merged revisions 167546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167546 | russell | 2009-01-07 16:20:31 -0600 (Wed, 07 Jan 2009) | 10 lines Merged revisions 167545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines Only try to close the file if one was actually opened ........ ................ * /, main/file.c: Merged revisions 167542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167542 | russell | 2009-01-07 16:05:29 -0600 (Wed, 07 Jan 2009) | 12 lines Merged revisions 167541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, #asterisk-dev ........ ................ 2009-01-07 18:32 +0000 [r167502] BJ Weschke * apps/app_followme.c, /: Merged revisions 167478 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167478 | bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines Answer the channel if it has not already been answered and we've already found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas ........ 2009-01-07 18:27 +0000 [r167491] Leif Madsen * /, configs/queues.conf.sample: Merged revisions 167477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 Jan 2009) | 8 lines Update queues.conf.sample documentation. Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so. (closes issue #14179) Reported by: CrashHD Tested by: CrashHD ........ 2009-01-07 17:46 +0000 [r167456] Russell Bryant * main/indications.c, /: Merged revisions 167442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167442 | russell | 2009-01-07 11:35:39 -0600 (Wed, 07 Jan 2009) | 12 lines Merged revisions 167432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) | 4 lines Treat an empty string the same way as a NULL country argument. In passing, simplify the handling of returning a default tone zone. ........ ................ 2009-01-07 14:41 +0000 [r167376] Leif Madsen * contrib/scripts/sip-friends.sql, /: Merged revisions 167373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167373 | lmadsen | 2009-01-07 09:26:19 -0500 (Wed, 07 Jan 2009) | 1 line Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is ........ 2009-01-06 21:38 +0000 [r167306] Mark Michelson * main/db.c, /: Merged revisions 167301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167301 | mmichelson | 2009-01-06 15:36:44 -0600 (Tue, 06 Jan 2009) | 16 lines Merged revisions 167299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) | 8 lines Use the correct variable when creating the format string (closes issue #14177) Reported by: nic_bellamy Patches: asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299) ........ ................ 2009-01-06 21:10 +0000 [r167268] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 167265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167265 | tilghman | 2009-01-06 15:02:33 -0600 (Tue, 06 Jan 2009) | 16 lines Merged revisions 167260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600 (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines Security fix AST-2009-001. ........ ................ ................ 2009-01-05 17:10 +0000 [r167182] Mark Michelson * /, channels/chan_sip.c: Merged revisions 167180 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan 2009) | 49 lines Merged revisions 167179 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines A couple of changes to T.38 SDP attribute handling There are some boolean attributes for T.38 such as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and T38FaxTranscodingJBIG. By simply being present, we should treat these as a "true" value. The current code, however, was requiring a 1 or 0 as the value of the attribute in order to parse it. This is due to the fact that there are some T.38 endpoints and gateways that also transmit this information incorrectly. This patch follows the "be liberal in what you accept and strict in what you send" philosophy by accepting both the correctly- and incorrectly-formatted attributes, but only sending information as it is supposed to be sent. It was also discovered that a particular type of T.38 gateway sends some non-standard T.38 SDP attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, it used T38MaxDatagram and T38FaxMaxRate respectively. We now will properly accept these attributes as well. Note that there are a lot of patches cited in the below commit message template. This is because the person who submitted these patches is an awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes issue #13976) Reported by: linulin Patches: chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648) Tested by: arcivanov ........ ................ 2009-01-05 16:46 +0000 [r167178] Tilghman Lesher * /, UPGRADE-1.6.txt: Merged revisions 167176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167176 | tilghman | 2009-01-05 10:44:47 -0600 (Mon, 05 Jan 2009) | 7 lines More clearly explain that quote marks are no longer necessary. (closes issue #13718) Reported by: davidw Patches: 20081020__bug13718.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ 2008-12-31 19:38 +0000 [r166957] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 166954 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166954 | tilghman | 2008-12-31 13:34:28 -0600 (Wed, 31 Dec 2008) | 12 lines Merged revisions 166953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines Also inherit the musiconhold class. (Closes #14153) Reported by: Jerry Geis, via the users list. Patch by: me (license 14) ........ ................ 2008-12-30 20:57 +0000 [r166910] Terry Wilson * phoneprov/polycom_line.xml, doc/realtimetext.txt, /, res/res_phoneprov.c, doc/sip-retransmit.txt, doc/tex/phoneprov.tex, res/res_http_post.c: Merged revisions 166908 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166908 | twilson | 2008-12-30 14:50:05 -0600 (Tue, 30 Dec 2008) | 2 lines Fix some svn:keywords ........ 2008-12-29 18:16 +0000 [r166863] Mark Michelson * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 166861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines Update app_queue to deal with the removal of AST_PBX_KEEPALIVE When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. ........ 2008-12-29 14:52 +0000 [r166858] Joshua Colp * channels/chan_sip.c: Per kpfleming add a note describing why you must never change the first element of peer_finding_info. 2008-12-28 15:16 +0000 [r166775] Russell Bryant * channels/misdn_config.c, /: Merged revisions 166773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166773 | russell | 2008-12-28 09:15:14 -0600 (Sun, 28 Dec 2008) | 12 lines Merged revisions 166772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines Use strncat() instead of an sprintf() in which source and target buffers overlap http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html ........ ................ 2008-12-24 01:15 +0000 [r166730] Steve Murphy * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c, main/pbx.c, /, main/features.c, apps/app_macro.c, include/asterisk/pbx.h: Merged revisions 166665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk This merged from trunk with no conflicts. I tested mostly the 'tired' cases, and for the most part ignored the tests for reconnecting and dialing in to fetch a parked call, after the first case. ................ r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ 2008-12-23 20:56 +0000 [r166698] Tilghman Lesher * include/asterisk/app.h, /, channels/chan_sip.c, main/app.c: Merged revisions 166696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines Allow semicolons and extended characters in user-specified SIP headers. (closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut ........ 2008-12-23 15:20 +0000 [r166571] Mark Michelson * main/channel.c, /: Merged revisions 166569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec 2008) | 20 lines Merged revisions 166568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ ................ 2008-12-23 04:34 +0000 [r166535] Tilghman Lesher * main/channel.c, /: Merged revisions 166533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) | 11 lines Merged revisions 166509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ ................ 2008-12-22 23:27 +0000 [r166440-166472] Mark Michelson * /, res/res_agi.c: Merged revisions 166470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166470 | mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11 lines Always use the value of the AGISIGHUP when running an AGI. Prior to this patch, the value of AGISIGUP was not always honored when set on a channel. (closes issue #13711) Reported by: fmueller Patches: 13711.patch uploaded by putnopvut (license 60) ........ * channels/chan_dahdi.c, /: Merged revisions 166382 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166382 | mmichelson | 2008-12-22 15:08:03 -0600 (Mon, 22 Dec 2008) | 44 lines Merged revisions 166380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks and autoservice It has been discovered that if a channel is locked prior to a call to ast_autoservice_stop, then it is likely that a deadlock will occur. The reason is that the call to ast_autoservice_stop has a check built into it to be sure that the thread running autoservice is not currently trying to manipulate the channel we are about to pull out of autoservice. The autoservice thread, however, cannot advance beyond where it currently is, though, because it is trying to acquire the lock of the channel for which autoservice is attempting to be stopped. The gist of all this is that a channel MUST NOT be locked when attempting to stop autoservice on the channel. In this particular case, the channel was locked by a call to ast_read. A call to ast_exists_extension led to autoservice being started and stopped due to the existence of dialplan switches. It may be that there are future commits which handle the same symptoms but in a different location, but based on my looks through the code, it is very rare to see a construct such as this one. (closes issue #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded by putnopvut (license 60) Tested by: rtrauntvein Review: http://reviewboard.digium.com/r/107/ ........ ................ 2008-12-22 21:46 +0000 [r166277-166438] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 166436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166436 | russell | 2008-12-22 15:45:28 -0600 (Mon, 22 Dec 2008) | 2 lines Cosmetic change - don't mix struct initializer styles. ........ * /, res/res_musiconhold.c: Merged revisions 166377 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166377 | russell | 2008-12-22 14:26:48 -0600 (Mon, 22 Dec 2008) | 2 lines Fix a bad typo. ........ * main/astobj2.c, /: Merged revisions 166342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166342 | russell | 2008-12-22 11:44:23 -0600 (Mon, 22 Dec 2008) | 2 lines Remove some error messages. This is the default handler that is valid to use. ........ * /, main/utils.c: Merged revisions 166317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166317 | russell | 2008-12-22 11:29:10 -0600 (Mon, 22 Dec 2008) | 10 lines Merged revisions 166297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines Fix up timeout handling in ast_carefulwrite(). ........ ................ * include/asterisk/utils.h, main/manager.c, /, main/utils.c: Merged revisions 166282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166282 | russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ ........ * /, res/res_musiconhold.c: Merged revisions 166273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166273 | russell | 2008-12-22 10:10:40 -0600 (Mon, 22 Dec 2008) | 7 lines Re-work ref count handling of MoH classes using astobj2 to resolve crashes. (closes issue #13566) Reported by: igorcarneiro Tested by: russell Review: http://reviewboard.digium.com/r/106/ ........ 2008-12-22 16:17 +0000 [r166275] Mark Michelson * /, funcs/func_timeout.c, main/file.c: Merged revisions 166267 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166267 | mmichelson | 2008-12-22 10:07:59 -0600 (Mon, 22 Dec 2008) | 17 lines Fix a file playback crash and explicitly initialize values in func_timeout.c A crash was brought up on the bugtracker. The first run through valgrind was full of legitimate complaints of uninitialized values in func_timeout when setting a response timeout. These were fixed but the crash persisted. A second run through showed the real problem. The reference counting used for filestreams was incorrect because there were some missing increments when a frame was read from a format module. (closes issue #14118) Reported by: blitzrage Patches: 14118v2.patch uploaded by putnopvut (license 60) Tested by: blitzrage ........ 2008-12-22 16:10 +0000 [r166272] Joshua Colp * main/dnsmgr.c, /: Merged revisions 166268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166268 | file | 2008-12-22 12:08:13 -0400 (Mon, 22 Dec 2008) | 7 lines Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port. (closes issue #13628) Reported by: pananix Patches: bug13628.patch uploaded by jpeeler (license 325) Tested by: file, blitzrage ........ 2008-12-22 14:19 +0000 [r166260] Russell Bryant * /, res/res_agi.c: Merged revisions 166258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166258 | russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines Remove AST_PBX_KEEPALIVE usage from res_agi. This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was for the AGI command, "asyncagi break". This patch removes this feature. Normally, a feature would not be removed like this. However, this code is broken and usage of it will result in a memory leak. Usage of this feature will make the AGI code return a result of AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed ownership of the channel. The channel thread will exit without destroying the channel. Unfortunately, _no_ thread has ownership of the channel at this point. There are a couple of serious problems here: 1) The only way to recover the caller is to issue a channel redirect. This will work, but this will be done with a masquerade, and the old ast_channel structure will be lost. 2) Until the channel redirect happens, there is no code servicing the channel. That means nothing is reading audio or handling events coming from the channel. This is very bad. The recommended way to get this same "break" functionality is to issue the redirect while the channel is still being handled by the AGI code. That way, there will be no memory leak, and there will be no period of time that the channel is not being serviced. ........ 2008-12-19 23:45 +0000 [r166098-166164] Mark Michelson * /, main/audiohook.c: Merged revisions 166162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166162 | mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6 lines Get rid of an extra space. I don't know how this crept back in when I had already fixed it earlier ........ * funcs/func_audiohookinherit.c: Switch documentation formats for func_audiohookinherit.c 1.6.1 does not have xml documentation, so I reverted to the old way here. * main/channel.c, funcs/func_audiohookinherit.c (added), /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 166092,166095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ ........ r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines Remove the verbatim tag from the author line I could have sworn I already did that before, though... ........ 2008-12-19 15:08 +0000 [r165892] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 165890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines Merged revisions 165889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ ................ 2008-12-18 Leif Madsen * Asterisk 1.6.1-beta4 released. 2008-12-18 21:57 +0000 [r165808] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 165797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165797 | tilghman | 2008-12-18 15:41:02 -0600 (Thu, 18 Dec 2008) | 15 lines Merged revisions 165767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines Add mutexes around accesses to the IMAP library interface. This prevents certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler ........ ................ 2008-12-18 21:47 +0000 [r165804] Russell Bryant * /, main/utils.c: Merged revisions 165801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008) | 19 lines Merged revisions 165796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines Make ast_carefulwrite() be more careful. This patch handles some additional cases that could result in partial writes to the file description. This was done to address complaints about partial writes on AMI. (issue #13546) (more changes needed to address potential problems in 1.6) Reported by: srt Tested by: russell Review: http://reviewboard.digium.com/r/99/ ........ ................ 2008-12-18 21:24 +0000 [r165794] Joshua Colp * apps/app_queue.c, channels/chan_oss.c, channels/chan_dahdi.c, channels/chan_misdn.c, /, channels/chan_sip.c, pbx/pbx_ael.c: Merged revisions 165792 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines Numerous documentation updates. (closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) ........ 2008-12-18 19:45 +0000 [r165728] Russell Bryant * apps/app_dial.c, main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 165723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ ........ 2008-12-18 19:36 +0000 [r165725] Mark Michelson * res/res_odbc.c, /: Merged revisions 165724 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165724 | mmichelson | 2008-12-18 13:34:33 -0600 (Thu, 18 Dec 2008) | 8 lines Fix crashes in res_odbc. The variable "class" was being set NULL just prior to being dereferenced in an ao2_link call. I have moved the setting of the variable to NULL until after the ao2_link call. ........ 2008-12-18 18:58 +0000 [r165664] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 165662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165662 | russell | 2008-12-18 12:54:47 -0600 (Thu, 18 Dec 2008) | 15 lines Merged revisions 165661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines Set the process group ID on the MOH process so that all children will get killed (closes issue #14099) Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645) ........ ................ 2008-12-18 18:47 +0000 [r165660] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 165658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines Fix 2 resource leaks and fix another pipe-to-comma conversion ........ 2008-12-18 17:59 +0000 [r165605-165606] Joshua Colp * /, channels/chan_sip.c: Merge in changes to return chan_sip to matching based on how it was previously done and how it is done in trunk. It will do name based for users and friends and IP based for peers. (closes issue #14107) Reported by: jsmith * main/rtp.c, /: Merged revisions 165599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines Merged revisions 165591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us. (closes issue #13545) Reported by: davidw ........ ................ 2008-12-18 16:48 +0000 [r165543] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 165541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165541 | tilghman | 2008-12-18 10:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines Fix reference counts of the class and add an assertion to the end. ........ 2008-12-17 21:48 +0000 [r165332] Mark Michelson * res/res_odbc.c, /: Merged revisions 165330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165330 | mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3 lines Fix a refcount leak in res_odbc ........ 2008-12-17 21:31 +0000 [r165329] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 165325 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165325 | tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines Oops, broke trunk ........ 2008-12-17 21:25 +0000 [r165324] Mark Michelson * apps/app_directory.c, apps/app_queue.c, apps/app_voicemail.c, /, res/res_realtime.c: Merged revisions 165318 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines Merged revisions 165255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ ................ 2008-12-17 21:22 +0000 [r165323] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 165319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) | 11 lines Merged revisions 165317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines Reverse the fix from issue #6176 and add proper handling for that issue. (Closes issue #13962, closes issue #13363) Fixed by myself (license 14) ........ ................ 2008-12-17 21:02 +0000 [r165279] Steve Murphy * /, utils/extconf.c: Merged revisions 165254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165254 | murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines This patch is here committed to satisfy the buildbot, who has a problem with the const. ........ 2008-12-17 20:02 +0000 [r165242] Terry Wilson * /, res/res_phoneprov.c: Merged revisions 165219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165219 | twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING. ........ 2008-12-17 19:54 +0000 [r165218] Joshua Colp * /, channels/chan_sip.c: Merged revisions 165216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by: chris-mac ........ 2008-12-17 17:56 +0000 [r165146] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 165142-165143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, 17 Dec 2008) | 10 lines Use the create_vm_state_from_user function in a place where it was not being used before. Also, I've moved the urgent folder check in messagecount() up a bit so that the flow is a bit better. This was something I noticed while taking a look at issue #13973, although I don't think this is the underlying cause of the issue. ........ r165143 | mmichelson | 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines And actually assign the function to a pointer... ........ 2008-12-17 05:53 +0000 [r165093] Steve Murphy * utils/conf2ael.c, pbx/ael/ael-test/ref.ael-vtest13, utils/check_expr.c, utils/Makefile, pbx/ael/ael-test/ref.ael-vtest17, /, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c: Merged revisions 165071 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk I might add here that in I tested the merged fixes from trunk in both 1.6.0 and 1.6.1 via both 'make' and ./runtests in the ael regression tests for all but DEBUG_CHANNEL_LOCKS, DEBUG_SCHEDULER, and CHANNEL_TRACE options. ........ r165071 | murf | 2008-12-16 22:04:56 -0700 (Tue, 16 Dec 2008) | 31 lines A possibly "horrible fix" for a "horribly broken" situation. As stuff shifts around in the asterisk code, the miscellaneous inclusions from the standalone stuff gets broken. There's no easy fix for this situation. I made sure that everything in utils builds without problem ***AND*** that aelparse runs the regressions correctly with the following make menuselect options both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to #undef all these features in the various utils native files; I guess I could do the same for the copied-in files, surrounded by STANDALONE ifdef. A standalone isn't going to care about threads, mutexes, etc. ........ 2008-12-16 23:07 +0000 [r164980] Mark Michelson * /, channels/chan_sip.c: Merged revisions 164978 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec 2008) | 15 lines Merged revisions 164977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines After looking through SIP registration code most of the day, this is one of the few things I could find that was just plain wrong. Even though it probably isn't possible for it to happen, it seems weird to have code that checks if a pointer is NULL and then immediately dereferences that pointer if it was NULL. ........ ................ 2008-12-16 22:52 +0000 [r164960] Jeff Peeler * /, apps/app_record.c: Merged revisions 164942 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164942 | jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines (closes issue #13669) Reported by: pj Delete file recording if recording terminated from a hangup. ........ 2008-12-16 21:40 +0000 [r164813-164884] Russell Bryant * /, main/utils.c: Merged revisions 164882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164882 | russell | 2008-12-16 15:39:15 -0600 (Tue, 16 Dec 2008) | 17 lines Merged revisions 164881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) | 9 lines Fix an issue where DEBUG_THREADS may erroneously report that a thread is exiting while holding a lock. If the last lock attempt was a trylock, and it failed, it will still be in the list of locks so that it can be reported. (closes issue #13219) Reported by: pj ........ ................ * /, apps/app_macro.c: Merged revisions 164877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines Merged revisions 164876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has been returned. This is a bug I noticed while looking at the code for app_macro. This return code means that another thread has assumed ownership of the channel and it can no longer be touched. (I hate this return code with a passion, by the way.) ........ ................ * main/manager.c, /: Merged revisions 164807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008) | 17 lines Merged revisions 164806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines Add "restart gracefully" to the AMI blacklist of CLI commands. "module unload" was already identified as a command that can not be used from the AMI. "restart gracefully" effectively unloads all modules, and will run in to the same problems. (closes issue #13894) Reported by: kernelsensei ........ ................ 2008-12-16 20:18 +0000 [r164805] Steve Murphy * main/pbx.c, /: Merged revisions 164801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164801 | murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines (closes issue #14076) Reported by: toc Tested by: murf OK, Well this issue has had its share of flip-flopping. I found the following: 1. the code in question, in ext_cmp1 in pbx.c, would not allow two extensions that vary only by any dashes contained within them, to be defined in the same context. 2. for input dialstrings, dashes are NOT ignored. So, skipping them when sorting patterns seemed a bit silly. Thus, you might declare ext 891 in a context, but if you try dialing 8-9-1, it will NOT match 891. So, I proposed to remove the code from ext_cmp1 to skip the spaces and dashes. Just kept us from declaring 891 and 8-9-1 in the same context, forcing users to generate otherwise uselessly obfuscated dialplan code to get the same effect. Then, I tried out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the same context! 2. You can't define 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have 891 match it! So, it appears that my proposal simply restores the pbx to behaving as it did in 1.4. ........ 2008-12-16 19:54 +0000 [r164799] Tilghman Lesher * contrib/scripts/safe_asterisk, /: Merged revisions 164798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164798 | tilghman | 2008-12-16 13:54:11 -0600 (Tue, 16 Dec 2008) | 4 lines Set up umask as a possible configuration option. (closes issue #13753) Reported by: irroot ........ 2008-12-16 17:18 +0000 [r164677-164739] Russell Bryant * include/asterisk/threadstorage.h, /, main/threadstorage.c: Merged revisions 164737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) | 22 lines Merged revisions 164736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ ................ * /, channels/chan_sip.c: Merged revisions 164675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines Merged revisions 164672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines Fix a memory leak related to the use of the "setvar" configuration option. The problem was that these variables were being appended to the list of vars on the sip_pvt every time a re-registration or re-subscription came in. Since it's just a waste of memory to put them there unless the request was an INVITE, then the fix is to check the request type before copying the vars. (closes issue #14037) Reported by: marvinek Tested by: russell ........ ................ 2008-12-16 15:47 +0000 [r164662] Joshua Colp * /, channels/chan_sip.c: Merged revisions 164659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164659 | file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself. (closes issue #13634) Reported by: performer ........ 2008-12-16 15:42 +0000 [r164658] Steve Murphy * main/pbx.c, /: Merged revisions 164648 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) | 13 lines Merged revisions 164634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 lines I added a sentence to clarify why - and ' ' are ignored in patterns as per bug 14076. Leif says he'll put some stuff about it in the extensions.conf sample, etc. ........ ................ 2008-12-16 15:02 +0000 [r164521-164625] Russell Bryant * /, apps/app_minivm.c: Merged revisions 164623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164623 | russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was not needed. (closes issue #14081) Reported by: pkempgen ........ * /, res/res_musiconhold.c: Merged revisions 164606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164606 | russell | 2008-12-16 08:31:02 -0600 (Tue, 16 Dec 2008) | 13 lines Merged revisions 164605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008) | 5 lines Don't try to change working directory if a directory was not configured. (closes issue #14089) Reported by: caspy ........ ................ * channels/chan_dahdi.c, /: Merged revisions 164602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008) | 7 lines Fix usage of the DAHDI_VMWI ioctl. (closes issue #14090) Reported by: alecdavis Patches: chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585) ........ * channels/chan_iax2.c, /: Merged revisions 164525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008) | 6 lines Open a timer before loading configuration so that the trunking configuration option will take effect. (closes issue #14082) Reported by: seandarcy ........ * channels/chan_iax2.c, /: Merged revisions 164522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164522 | russell | 2008-12-15 16:22:43 -0600 (Mon, 15 Dec 2008) | 4 lines Fix log message to refer to the generic timing interface, not DAHDI specifically (inspired by issue #14082) ........ * main/frame.c, /: Merged revisions 164519 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164519 | russell | 2008-12-15 15:53:30 -0600 (Mon, 15 Dec 2008) | 7 lines Make sure we handle a uint32_t payload in ast_frdup() (closes issue #14080) Reported by: fnordian Patches: frame.patch uploaded by fnordian (license 110) ........ 2008-12-15 19:54 +0000 [r164421-164425] Mark Michelson * /, include/asterisk/pbx.h: Merged revisions 164423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164423 | mmichelson | 2008-12-15 13:53:29 -0600 (Mon, 15 Dec 2008) | 11 lines Merged revisions 164422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines Add the deadlock note to ast_spawn_extension as well ........ ................ * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged revisions 164419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec 2008) | 12 lines Merged revisions 164416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future ........ ................ 2008-12-15 18:27 +0000 [r164355] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 164349 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164349 | tilghman | 2008-12-15 12:09:58 -0600 (Mon, 15 Dec 2008) | 4 lines When querying for the structure of the CDR table, remove the schema, if it exists. (Closes issue #14058) ........ 2008-12-15 18:14 +0000 [r164314-164353] Joshua Colp * /, channels/chan_sip.c: Merged revisions 164351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) | 13 lines Merged revisions 164350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines Do not try to unlock a non-existant channel if the transfer fails. (closes issue #13800) Reported by: dwagner Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608) ........ ................ * /, main/file.c: Merged revisions 164312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164312 | file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module. (closes issue #14079) Reported by: elguero ........ 2008-12-15 16:32 +0000 [r164276-164300] Russell Bryant * main/channel.c, /, main/features.c: Merged revisions 164203 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 164272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164272 | russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines When a reload is issued, always process the configuration for dundi.conf. The reason is that a reload can be used to refresh DNS lookups for defined peers. Even if the config file hasn't changed, we want to process it for that purpose. (closes issue #13776) Reported by: kombjuder ........ 2008-12-15 16:18 +0000 [r164273-164274] Mark Michelson * apps/app_queue.c, /: Merged revisions 164270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164270 | mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4 lines Fix a compile warning and a logic error that could have been bad for non-realtime queues ........ * apps/app_queue.c, /: Merged revisions 164268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines Fix up a few issues with regards to queues * Fix reference counting used in the __queues_show function * Add code to be sure that the "queue show" command does not print information for a realtime queue which has been deleted from the backend * Add a missing unref to the realtime queue loading function for the case where a queue is in the module's container but has been deleted from the realtime backend (closes issue #14033) Reported by: cristiandimache Patches: 14033.patch uploaded by putnopvut (license 60) Tested by: cristiandimache ........ 2008-12-15 15:50 +0000 [r164266] Joshua Colp * /, configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, configure.ac: Merged revisions 164257 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 | file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy ........ 2008-12-13 01:01 +0000 [r163914] Joshua Colp * apps/app_chanspy.c, /: Merged revisions 163912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163912 | file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines Only detach and destroy the whisper audiohooks if they are actually in use. ........ 2008-12-13 00:08 +0000 [r163875] Terry Wilson * apps/app_queue.c, /: Merged revisions 163873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163873 | twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered. (closes issue #14034) Reported by: cristiandimache Tested by: otherwiseguy, cristiandimache ........ 2008-12-12 23:08 +0000 [r163830] Russell Bryant * /: Merged revisions 163829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ 2008-12-12 22:05 +0000 [r163764] Tilghman Lesher * main/asterisk.c, main/editline/read.c, /: Merged revisions 163762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163762 | tilghman | 2008-12-12 16:04:26 -0600 (Fri, 12 Dec 2008) | 14 lines Merged revisions 163761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a pointer inside editline to look back to asterisk.c, so others don't spend as much time as I did looking (in the wrong place) for the appropriate function. Reported by: ZX81, via the #asterisk-users channel Fixed by: me (license 14) ........ ................ 2008-12-12 19:58 +0000 [r163715] Steve Murphy * channels/chan_dahdi.c, /: Merged revisions 163675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163675 | murf | 2008-12-12 12:16:32 -0700 (Fri, 12 Dec 2008) | 1 line demote always-appearing debug message (for certain boards) to ast_debug lev 3 msg instead ........ 2008-12-12 18:53 +0000 [r163656-163672] Russell Bryant * main/tcptls.c, /, channels/chan_sip.c: Merged revisions 163670 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines Rename a number of tcptls_session variables. There are no functional changes here. The name "ser" was used in a lot of places. However, it is a relic from when the struct was a server_instance, not a session_instance. It was renamed since it represents both a server or client connection. ........ * /, channels/chan_sip.c: Merged revisions 163667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163667 | russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines Fix a small race condition in sip_tcp_locate(). We must increase the reference count on the tcptls_session _before_ unlocking the thread list. ........ * /, channels/chan_sip.c: Merged revisions 163642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163642 | russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines Resolve crashes when using SIP TCP/TLS with qualify. The problem was a reference count error on the tcptls_session structure. (closes issue #13989) Reported by: Nugget ........ 2008-12-12 18:19 +0000 [r163640] Joshua Colp * /, channels/chan_sip.c: Merged revisions 163629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found. (closes issue #13811) Reported by: pj ........ 2008-12-12 17:26 +0000 [r163624] Michiel van Baak * res/res_monitor.c, /: Merged revisions 163612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163612 | mvanbaak | 2008-12-12 18:22:47 +0100 (Fri, 12 Dec 2008) | 7 lines Document default Monitor file location. (closes issue #14065) Reported by: kshumard Patches: res_monitor.documentation.patch.txt uploaded by kshumard (license 92) ........ 2008-12-12 16:57 +0000 [r163581] Joshua Colp * main/channel.c, /, channels/chan_sip.c: Merged revisions 163579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj ........ 2008-12-12 14:48 +0000 [r163514-163515] Russell Bryant * main/channel.c, main/autoservice.c, /, include/asterisk/channel.h: Merged revisions 163449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 163512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163512 | russell | 2008-12-12 08:44:06 -0600 (Fri, 12 Dec 2008) | 13 lines Merged revisions 163511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) | 5 lines Specify uint32_t for variables storing a CRC32 so that it is actually 32 bits on 64-bit machines, as well. (inspired by issue #13879) ........ ................ 2008-12-11 23:48 +0000 [r163386] Tilghman Lesher * main/asterisk.c, /: Merged revisions 163384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163384 | tilghman | 2008-12-11 17:38:56 -0600 (Thu, 11 Dec 2008) | 16 lines Merged revisions 163383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal is messed up. By intercepting those events with a signal handler in the remote console, we can avoid those issues. (closes issue #13464) Reported by: tzafrir Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ ................ 2008-12-11 22:52 +0000 [r163319] Matt Nicholson * /, pbx/pbx_dundi.c: Merged revisions 163317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163317 | mnicholson | 2008-12-11 16:49:59 -0600 (Thu, 11 Dec 2008) | 16 lines Merged revisions 163316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes issue #13819) Reported by: adomjan Patches: pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) dundi_clearecache3.diff uploaded by mnicholson (license 96) Tested by: adomjan ........ ................ 2008-12-11 21:50 +0000 [r163252-163256] Russell Bryant * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions 163254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163254 | russell | 2008-12-11 15:48:08 -0600 (Thu, 11 Dec 2008) | 16 lines Merged revisions 163253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines Fix some observed slowdowns in dialplan processing. The change is to remove autoservice usage from dialplan functions that do not need it because they do not perform operations that potentially block. (closes issue #13940) Reported by: tbelder ........ ................ * /, res/res_timing_pthread.c: Merged revisions 163241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163241 | russell | 2008-12-11 15:21:31 -0600 (Thu, 11 Dec 2008) | 8 lines Fix a problem where continuous mode will get inadvertently get turned off if set_rate() is used while continuous mode was already turned on. (closes issue #13738) Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix (license 547) ........ 2008-12-11 21:00 +0000 [r163214] Mark Michelson * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 163213 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163213 | mmichelson | 2008-12-11 14:57:44 -0600 (Thu, 11 Dec 2008) | 9 lines Add an option to voicemail.conf to allow urgent messages to be forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) ........ 2008-12-11 20:10 +0000 [r163173] Russell Bryant * main/channel.c, /: Merged revisions 163171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163171 | russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza ........ 2008-12-11 20:05 +0000 [r163170] Tilghman Lesher * /, configure, configure.ac: Merged revisions 163168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163168 | tilghman | 2008-12-11 14:02:35 -0600 (Thu, 11 Dec 2008) | 5 lines Sometimes even Linux needs -lm to link libtonezone, such as when libtonezone is compiled statically. (closes issue #13887) Reported by: tzafrir ........ 2008-12-11 17:16 +0000 [r163100] Russell Bryant * /, main/features.c: Merged revisions 163094 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008) | 19 lines Merged revisions 163092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines Fix an issue that made it so you could only have a single caller executing a custom feature at a time. This was especially problematic when custom features ran for any appreciable amount of time. The fix turned out to be quite simple. The dynamic features are now stored in a read/write list instead of a list using a mutex. (closes issue #13478) Reported by: neutrino88 Fix suggested by file ........ ................ 2008-12-11 16:54 +0000 [r163091] Tilghman Lesher * /, res/res_agi.c: Merged revisions 163089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163089 | tilghman | 2008-12-11 10:52:24 -0600 (Thu, 11 Dec 2008) | 13 lines Merged revisions 163088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) | 6 lines Don't wait forever, if there's a specified recording timeout. (closes issue #13885) Reported by: bamby Patches: res_agi.c.patch uploaded by bamby (license 430) ........ ................ 2008-12-11 16:49 +0000 [r163083-163087] Mark Michelson * apps/app_queue.c, /: Merged revisions 163085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163085 | mmichelson | 2008-12-11 10:47:34 -0600 (Thu, 11 Dec 2008) | 12 lines Merged revisions 163084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines Revert this cast to long. Using time_t here causes build failures on a FreeBSD 32-bit build. ........ ................ * apps/app_queue.c, /: Merged revisions 163081 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec 2008) | 22 lines Merged revisions 163080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines Fix a potential crash due to unsafe datastore handling. This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek ........ ................ 2008-12-11 15:07 +0000 [r163006] Joshua Colp * /, channels/chan_sip.c: Merged revisions 162997 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162997 | file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it. (closes issue #13525) Reported by: pj ........ 2008-12-10 23:13 +0000 [r162949] Tilghman Lesher * main/pbx.c, /: Merged revisions 162922,162930 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162922 | tilghman | 2008-12-10 16:48:09 -0600 (Wed, 10 Dec 2008) | 7 lines Checking global variables here actually overwrote the previous substitution by channel variables, and in any case, was redundant; pbx_substitute_variables_helper ALREADY does substitution for global variables. (closes issue #13327) Reported by: pj ........ r162930 | tilghman | 2008-12-10 17:01:14 -0600 (Wed, 10 Dec 2008) | 2 lines Previously missing line, now the substitution works correctly ........ 2008-12-10 22:54 +0000 [r162896-162929] Jeff Peeler * /, res/res_musiconhold.c: Merged revisions 162927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162927 | jpeeler | 2008-12-10 16:53:34 -0600 (Wed, 10 Dec 2008) | 11 lines Merged revisions 162926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check. Pointed out by mmichelson, thanks! ........ ................ * /, res/res_musiconhold.c: Merged revisions 162891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162891 | jpeeler | 2008-12-10 16:11:46 -0600 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines (closes issue #13229) Reported by: clegall_proformatique Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams. ........ ................ 2008-12-10 19:05 +0000 [r162741-162807] Joshua Colp * /, channels/chan_sip.c: Merged revisions 162805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI. (closes issue #12560) Reported by: vsauer Patches: patch001.diff uploaded by ramonpeek (license 266) ........ ................ * /, channels/chan_sip.c: Merged revisions 162739 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0. (closes issue #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded by hjourdain (license 583) ........ ................ 2008-12-10 16:39 +0000 [r162666-162669] Mark Michelson * doc/tex/misdn.tex, /: Merged revisions 162667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162667 | mmichelson | 2008-12-10 10:39:10 -0600 (Wed, 10 Dec 2008) | 16 lines Merged revisions 162659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines Add missing documentation to misdn.txt (closes issue #14052) Reported by: festr Patches: misdn.txt.patch uploaded by festr (license 443) ........ ................ * /, channels/chan_sip.c: Merged revisions 162664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec 2008) | 19 lines Merged revisions 162663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines Revert fix for issue 13570. It has caused more problems than it helped to fix. (closes issue #13783) Reported by: navkumar (closes issue #14025) Reported by: ffs ........ ................ 2008-12-10 16:08 +0000 [r162622-162658] Joshua Colp * main/rtp.c, /: Merged revisions 162656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change. (closes issue #12983) Reported by: vt Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520) ........ ................ * /, channels/chan_sip.c: Merged revisions 162619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162619 | file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server. (closes issue #13633) Reported by: performer ........ 2008-12-10 11:37 +0000 [r162585] Michiel van Baak * /, res/snmp/agent.c: Merged revisions 162583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162583 | mvanbaak | 2008-12-10 12:34:09 +0100 (Wed, 10 Dec 2008) | 5 lines Make res_snmp.so compile on OpenBSD. OpenBSD uses an old version of gcc which throws an error if you use a macro that's not #defined ........ 2008-12-09 23:45 +0000 [r162490] Mark Michelson * include/asterisk/stringfields.h, /: Merged revisions 162488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162488 | kpfleming | 2008-12-09 17:41:02 -0600 (Tue, 09 Dec 2008) | 1 line it does help if the compiler attribute syntax is correct ........ 2008-12-09 23:12 +0000 [r162472] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 162466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162466 | tilghman | 2008-12-09 17:10:34 -0600 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ ................ 2008-12-09 22:34 +0000 [r162416] Russell Bryant * main/asterisk.c, include/asterisk/utils.h, /, main/utils.c: Merged revisions 162414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) | 16 lines Merged revisions 162413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ ................ 2008-12-09 22:02 +0000 [r162372] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 162355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162355 | tilghman | 2008-12-09 15:57:09 -0600 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines We appear to have documented tz= in the [general] section of voicemail.conf, without actually having implemented it. Oops. (Reported by Olivier on the -users list) ........ ................ 2008-12-09 21:18 +0000 [r162344] Joshua Colp * /, apps/app_directed_pickup.c: Merged revisions 162342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162342 | file | 2008-12-09 17:16:37 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing. (closes issue #14005) Reported by: ddl ........ ................ 2008-12-09 21:03 +0000 [r162302] Russell Bryant * /, apps/app_meetme.c: Merged revisions 162291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines Merged revisions 162286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ ................ 2008-12-09 20:48 +0000 [r162278] Joshua Colp * /, apps/app_festival.c: Merged revisions 162275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162275 | file | 2008-12-09 16:46:11 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines Fix double declaration of 'x' on the PPC platform. (closes issue #14038) Reported by: ffloimair ........ ................ 2008-12-09 20:47 +0000 [r162277] Steve Murphy * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162271 | murf | 2008-12-09 13:40:31 -0700 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this. ........ ................ 2008-12-09 20:31 +0000 [r162269] Mark Michelson * main/pbx.c, /: Merged revisions 162266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec 2008) | 14 lines Merged revisions 162265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines If we fail to start a thread for the pbx to run in, we need to be sure to decrease the number of active calls on the system. This fix may relate to ABE-1713, but it is not certain yet. ........ ................ 2008-12-09 19:52 +0000 [r162202-162207] Joshua Colp * main/rtp.c, /: Merged revisions 162205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines Merged revisions 162204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment. (closes issue #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file (license 11) Tested by: ip-rob, bujones ........ ................ * main/rtp.c, /: Merged revisions 162197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines Take video into account when early bridging RTP. (closes issue #13535) Reported by: davidw ........ ................ 2008-12-09 18:49 +0000 [r162082-162142] Steve Murphy * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162140 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162140 | murf | 2008-12-09 11:35:35 -0700 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 line Previous fix used ast_malloc and ast_copy_string and messed up the standalone stuff. Fixed. ........ ................ * res/ael/ael.flex, res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c: Merged revisions 162079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) | 53 lines Merged revisions 162013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! ........ ................ 2008-12-09 16:50 +0000 [r161963-162018] Russell Bryant * /, apps/app_disa.c: Merged revisions 162016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162016 | russell | 2008-12-09 10:47:39 -0600 (Tue, 09 Dec 2008) | 13 lines Merged revisions 162014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines Allow DISA to handle extensions that start with #. (closes issue #13330) Reported by: jcovert ........ ................ * /, main/app.c: Merged revisions 161951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161951 | russell | 2008-12-09 08:57:39 -0600 (Tue, 09 Dec 2008) | 23 lines Merged revisions 161948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines Fix a problem with GROUP() settings on a masquerade. The previous code carried over group settings from the old channel to the new one. However, it did nothing with the group settings that were already on the new channel. This patch removes all group settings that already existed on the new channel. I have a more complicated version of this patch which addresses only the most blatant problem with this, which is that a channel can end up with multiple group settings in the same category. However, I could not think of a use case for keeping any of the group settings from the old channel, so I went this route for now. (closes AST-152) ........ ................ 2008-12-08 20:55 +0000 [r161835] Joshua Colp * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: Merged revisions 161830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161830 | file | 2008-12-08 16:53:50 -0400 (Mon, 08 Dec 2008) | 2 lines Update autosupport script with a few changes. ........ 2008-12-08 18:52 +0000 [r161792] Tilghman Lesher * main/manager.c, /: Merged revisions 161790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161790 | tilghman | 2008-12-08 12:49:50 -0600 (Mon, 08 Dec 2008) | 6 lines Allocate enough space initially for the message. (closes issue #14027) Reported by: junky Patches: M14027.diff uploaded by junky (license 177) ........ 2008-12-08 18:49 +0000 [r161729-161789] Joshua Colp * main/pbx.c, /: Merged revisions 161787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161787 | file | 2008-12-08 14:47:32 -0400 (Mon, 08 Dec 2008) | 6 lines Fix a regression introduced when the PBX timeouts were converted to milliseconds. collect_digits now gets milliseconds fed to it, not seconds. (closes issue #14012) Reported by: dveiga Patches: 14012.patch uploaded by bkruse (license 132) ........ * /, channels/chan_sip.c: Merged revisions 161726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) | 13 lines Merged revisions 161725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines Make the usereqphone option work again. (closes issue #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff uploaded by mmaguire (license 571) ........ ................ 2008-12-08 17:24 +0000 [r161722] Matt Nicholson * /, channels/chan_sip.c: Merged revisions 161721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161721 | mnicholson | 2008-12-08 11:23:41 -0600 (Mon, 08 Dec 2008) | 7 lines Fix a crash that can occur on a transfer in chan_sip when attempting to collect rtp stats. (closes issue #13956) Reported by: chris-mac Tested by: chris-mac ........ 2008-12-05 23:29 +0000 [r161496] Mark Michelson * apps/app_stack.c, /: Merged revisions 161493 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161493 | mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 lines If the autoloop flag is set on a channel, then we need to add 1 to the priority when checking if the extension exists. Otherwise, gosubs will fail. This was discovered when investigating an asterisk-users mailing list post made by Gary Hawkins. ........ 2008-12-05 21:16 +0000 [r161352-161429] Sean Bright * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 161427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161427 | seanbright | 2008-12-05 16:08:43 -0500 (Fri, 05 Dec 2008) | 22 lines Merged revisions 161426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ ................ ................ * apps/app_voicemail.c, /: Merged revisions 161349-161350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, 05 Dec 2008) | 5 lines When using IMAP_STORAGE, it's important to convert bare newlines (\n) in emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed by Mark M. on IRC. ........ r161350 | seanbright | 2008-12-05 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines Use ast_free() instead of free(), pointed out by eliel on IRC. ........ 2008-12-05 14:18 +0000 [r161285-161290] Russell Bryant * main/pbx.c, /: Merged revisions 161288 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161288 | russell | 2008-12-05 08:16:24 -0600 (Fri, 05 Dec 2008) | 10 lines Merged revisions 161287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) | 2 lines Fix a NULL format string warning found by buildbot. ........ ................ * /, apps/app_minivm.c: Merged revisions 161252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161252 | russell | 2008-12-05 07:46:01 -0600 (Fri, 05 Dec 2008) | 2 lines Resolve a compiler warning from buildbot about a NULL format string. ........ 2008-12-05 05:42 +0000 [r161182] Tilghman Lesher * main/config.c, /: Merged revisions 161181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161181 | tilghman | 2008-12-04 23:41:41 -0600 (Thu, 04 Dec 2008) | 11 lines The first file should have a blank config filename in the structure, so that when a save occurs to a different filename, everything goes to the alternate filename, instead of appending to the original. This is important for the AMI command UpdateConfig. (closes issue #13301) Reported by: trevo Patches: 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14) 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ 2008-12-05 02:52 +0000 [r161149] Sean Bright * apps/app_voicemail.c, /: Merged revisions 161147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161147 | seanbright | 2008-12-04 21:47:54 -0500 (Thu, 04 Dec 2008) | 3 lines Check the return value of fread/fwrite so the compiler doesn't complain. Only a problem when IMAP_STORAGE is enabled. ........ 2008-12-04 18:37 +0000 [r161016] Jeff Peeler * main/rtp.c, /: Merged revisions 161014 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) | 17 lines Merged revisions 161013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines (closes issue #13835) Reported by: matt_b Tested by: jpeeler This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure. Closes AST-142. ........ ................ 2008-12-04 16:48 +0000 [r160947] Mark Michelson * /, main/callerid.c: Merged revisions 160945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160945 | mmichelson | 2008-12-04 10:45:06 -0600 (Thu, 04 Dec 2008) | 23 lines Merged revisions 160943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines Fix a callerid parsing issue. If someone formatted callerid like the following: "name " (including the quotation marks), then the parts would be parsed as name: "name number: number This is because the closing quotation mark was not discovered since the number and everything after was parsed out of the string earlier. Now, there is a check to see if the closing quote occurs after the number, so that we can know if we should strip off the opening quote on the name. Closes AST-158 ........ ................ 2008-12-04 01:41 +0000 [r160858-160859] Richard Mudgett * funcs/func_callerid.c, /: Merged revisions 160856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160856 | rmudgett | 2008-12-03 19:36:39 -0600 (Wed, 03 Dec 2008) | 1 line Jcolp pointed out that num will also match number ........ * funcs/func_callerid.c, /: Merged revisions 160854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160854 | rmudgett | 2008-12-03 19:14:22 -0600 (Wed, 03 Dec 2008) | 4 lines * Found a couple more places where num/number needed to be done so 1.4 upgraders will not have problems. * Added curly braces and minor tweaks. ........ 2008-12-03 22:02 +0000 [r160811] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 160791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160791 | tilghman | 2008-12-03 15:58:21 -0600 (Wed, 03 Dec 2008) | 9 lines Merged revisions 160770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ ................ 2008-12-03 21:40 +0000 [r160766] Steve Murphy * funcs/func_callerid.c, /: Merged revisions 160760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160760 | murf | 2008-12-03 14:09:15 -0700 (Wed, 03 Dec 2008) | 23 lines Merged revisions 160703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines (closes issue #13597) Reported by: john8675309 Patches: patch.13597 uploaded by murf (license 17) Tested by: murf, john8675309 This patch causes the setcid func to update the CDR clid after setting the channel field. I also notice that in trunk, the num/number of 1.4 is left out; I decided to include the option to use either in trunk, so as not to have 1.4 upgraders not to have problems. ........ ................ 2008-12-03 20:36 +0000 [r160702] Jason Parker * main/manager.c, /: Merged revisions 160699-160700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160699 | qwell | 2008-12-03 14:32:20 -0600 (Wed, 03 Dec 2008) | 7 lines Fix typo when ListCategories returns none. (closes issue #13994) Reported by: mika Patches: ListCategoriesActionPatch.diff uploaded by mika (license 624) ........ r160700 | qwell | 2008-12-03 14:35:36 -0600 (Wed, 03 Dec 2008) | 1 line Another place this is missing ........ 2008-12-03 19:49 +0000 [r160665] Eliel C. Sardanons * /, channels/iax2-provision.c: Merged revisions 160663 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec 2008) | 13 lines - iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module. - Move the code to start using the LIST macros. Review: http://reviewboard.digium.com/r/72 (closes issue #13232) Reported by: eliel Patches: iax2-provision.patch.txt uploaded by eliel (license 64) (with minor changes pointed by Mark Michelson on review board) Tested by: eliel ........ 2008-12-03 18:42 +0000 [r160628] Mark Michelson * apps/app_queue.c, apps/app_stack.c, apps/app_dial.c, /: Merged revisions 160626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines Add some safety measures when using gosub, especially when using the options for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur ........ 2008-12-03 17:41 +0000 [r160561] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 160559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160559 | tilghman | 2008-12-03 11:38:59 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines If an entry is added to the directory during a scan when another entry expires, then that new entry will not be processed promptly, but must wait for either a future entry to start or a current entry's retry to occur. If no other entries exist in the directory (other than the new entries) when a bunch expire, then the new entries must wait until another new entry is added to be processed. This was a rather weird race condition, really. Fixes AST-147. ........ ................ 2008-12-03 17:10 +0000 [r160557] Mark Michelson * apps/app_queue.c, /: Merged revisions 160555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160555 | mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 lines When investigating issue #13548, I found that gosub handling in app_queue was just completely wrong, mostly because the channel operations being performed were being done on the incorrect channel. With this set of changes, a gosub will correctly run on the answering queue member's channel. There are still crash issues which occur if there are dialplan syntax errors, so I cannot yet close the referenced issue. ........ 2008-12-03 17:02 +0000 [r160483-160554] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 160552 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160552 | tilghman | 2008-12-03 11:01:03 -0600 (Wed, 03 Dec 2008) | 11 lines Merged revisions 160551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) | 4 lines Don't start scanning the directory until all modules are loaded, because some required modules (channels, apps, functions) may not yet be in memory yet. Fixes AST-149. ........ ................ * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ ................ 2008-12-02 18:05 +0000 [r160329-160339] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) | 1 line remove duplicate comment that I accidentally merged ........ * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) | 7 lines (closes issue #13786) Reported by: tzafrir Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters: http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 ........ 2008-12-02 18:03 +0000 [r160234-160325] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines Merged revisions 160297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ ................ * include/asterisk/stringfields.h, apps/app_voicemail.c, main/cli.c, main/pbx.c, main/frame.c, /, channels/chan_features.c: Merged revisions 160208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ 2008-12-01 23:53 +0000 [r160175] Sean Bright * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged revisions 160170-160172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec 2008) | 1 line Pay attention to the return value of system(), even if we basically ignore it. ................ r160171 | seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 line Silence a build warning. (chan_phone.c:810: warning: value computed is not used) ................ r160172 | seanbright | 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged revisions 159976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines Get rid of the useless format string and argument in the Bogus/ manager channelname. Noted by kpfleming and name Bogus/manager suggested by eliel ........ ................ 2008-12-01 Tilghman Lesher * Released 1.6.1-beta3 2008-12-01 21:46 +0000 [r160101] Tilghman Lesher * /, configure, configure.ac: Merged revisions 160097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or bad things happen. ........ 2008-12-01 17:45 +0000 [r160006] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 160004 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160004 | russell | 2008-12-01 11:34:31 -0600 (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they both have the potential to send control frames in the middle of call setup. We have to wait until we have received a message back from the remote end before we try to send any more frames. Otherwise, the remote end will consider it invalid, and we'll get stuck in an INVAL/VNAK storm. ........ ................ 2008-12-01 16:06 +0000 [r159975] Michiel van Baak * main/manager.c, /: Merged revisions 159898 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) | 11 lines Merged revisions 159897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) | 4 lines make manager compile on OpenBSD. The last (10th) argument to ast_channel_alloc here should be a pointer and NULL is not really a pointer. ........ ................ 2008-12-01 14:57 +0000 [r159920] Russell Bryant * .cleancount, /: Merged revisions 159911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 159900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) | 2 lines Force a "make clean" to avoid a bizarre build issue ... ........ ................ 2008-11-29 18:34 +0000 [r159854] Tilghman Lesher * /, apps/app_readexten.c: Merged revisions 159853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159853 | tilghman | 2008-11-29 12:33:18 -0600 (Sat, 29 Nov 2008) | 2 lines Allow the '#' sign to exist within an extension (inspired by issue #13330) ........ 2008-11-29 18:16 +0000 [r159851] Kevin P. Fleming * channels/chan_iax2.c, cdr/cdr_tds.c, include/asterisk/logger.h, include/asterisk/res_odbc.h, channels/chan_misdn.c, include/asterisk/astmm.h, include/asterisk/lock.h, utils/extconf.c, makeopts.in, main/dns.c, funcs/Makefile, include/asterisk/stringfields.h, include/asterisk/utils.h, include/asterisk/devicestate.h, /, include/asterisk/dundi.h, configure.ac, utils/astman.c, include/asterisk/cli.h, include/asterisk/channel.h, include/asterisk/manager.h, res/res_config_sqlite.c, utils/conf2ael.c, utils/frame.c, channels/misdn_config.c, main/ast_expr2.c, Makefile, main/srv.c, include/asterisk/compat.h, configure, channels/misdn/ie.c, include/asterisk/module.h, main/features.c, include/asterisk/linkedlists.h, main/logger.c, main/event.c, include/asterisk/dlinkedlists.h, include/asterisk/strings.h, utils/check_expr.c, channels/chan_vpb.cc, channels/chan_sip.c, main/Makefile, include/asterisk/enum.h, channels/chan_agent.c, main/utils.c, include/jitterbuf.h: Merged revisions 159818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ 2008-11-26 19:58 +0000 [r159561] Mark Michelson * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines Add some necessary hangup commands in the case that forwarding a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw ........ 2008-11-26 19:17 +0000 [r159535] Kevin P. Fleming * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 159534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov 2008) | 11 lines Merged revisions 159476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov 2008) | 7 lines simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode ........ ................ 2008-11-26 18:38 +0000 [r159477] Tilghman Lesher * main/udptl.c, /: Merged revisions 159475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 | tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines If the config file does not exist, then the first use crashes Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage ........ 2008-11-26 14:59 +0000 [r159438] Mark Michelson * /, channels/chan_agent.c: Merged revisions 159437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 | mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10 lines Don't allow for configuration options to overwrite options set via channel variables on a reload. (closes issue #13921) Reported by: davidw Patches: 13921.patch uploaded by putnopvut (license 60) Tested by: davidw ........ 2008-11-26 03:19 +0000 [r159403] Jeff Peeler * /, main/features.c: Merged revisions 159402 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159402 | jpeeler | 2008-11-25 21:18:01 -0600 (Tue, 25 Nov 2008) | 3 lines Always parse arguments in park_call_exec so that app_args is valid. This prevents a crash when executing Park from the dialplan with no arguments. ........ 2008-11-25 23:27 +0000 [r159375] Steve Murphy * channels/chan_iax2.c, main/cdr.c, /: Merged revisions 159360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines (closes issue #12694) Reported by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX fix. the change to cdr.c allows no-answer to percolate up into CDR's, and feels like the right place to locate this fix; if BUSY is done here, no-answer should be, too. ........ ................ 2008-11-25 21:58 +0000 [r159249-159280] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 159276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ ................ * channels/chan_iax2.c, /: Merged revisions 159247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600 (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair ........ ................ ................ 2008-11-25 16:21 +0000 [r159095] Terry Wilson * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 | twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines Add missing variable declaration for PPC code ........ 2008-11-25 05:05 +0000 [r159053] Tilghman Lesher * channels/xpmr/xpmr.c, apps/app_rpt.c, channels/chan_usbradio.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 159050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159050 | tilghman | 2008-11-24 23:02:11 -0600 (Mon, 24 Nov 2008) | 10 lines Merged revisions 159025 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ ................ 2008-11-25 03:51 +0000 [r158993] Terry Wilson * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) | 2 lines Make chan_usbradio compile under dev mode ........ 2008-11-25 00:41 +0000 [r158894-158927] Matt Nicholson * apps/app_queue.c, /, UPGRADE.txt: Merged revisions 158924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158924 | mnicholson | 2008-11-24 18:05:41 -0600 (Mon, 24 Nov 2008) | 6 lines Make the Join event from app_queue use CallerIDNum insead of CallerID for indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw ........ * /, main/file.c: Merged revisions 158925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158925 | mnicholson | 2008-11-24 18:19:55 -0600 (Mon, 24 Nov 2008) | 2 lines Fix compiling in dev mode. ........ * include/asterisk/manager.h, main/manager.c, /, res/res_agi.c: Merged revisions 158876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158876 | mnicholson | 2008-11-24 15:56:22 -0600 (Mon, 24 Nov 2008) | 7 lines Added EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes issue #13873) Reported by: fnordian Patches: ami_agievent.patch uploaded by fnordian (license 110) ........ 2008-11-24 21:53 +0000 [r158861] Tilghman Lesher * main/dsp.c, /: Merged revisions 158857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158857 | tilghman | 2008-11-24 15:52:34 -0600 (Mon, 24 Nov 2008) | 3 lines Add a bit of documentation (thanks, I-MOD) on what the silence threshold constant actually does and what values are valid for it. ........ 2008-11-24 21:44 +0000 [r158855] Matt Nicholson * /, main/file.c: Merged revisions 158851 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158851 | mnicholson | 2008-11-24 15:27:26 -0600 (Mon, 24 Nov 2008) | 6 lines Make ast_streamfile() check the result of ast_openstream() before doing anything with it. (closes issue #13955) Reported by: chris-mac ........ 2008-11-22 17:00 +0000 [r158689-158701] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 158694 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008) | 8 lines dont send reorder tone after a device is hungup if a dialout is abandoned or failed. Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box (closes issue #13948) Reported by: wedhorn Patches: switch.diff uploaded by wedhorn (license 30) ........ * /, channels/chan_skinny.c: Merged revisions 158688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158688 | mvanbaak | 2008-11-22 17:06:38 +0100 (Sat, 22 Nov 2008) | 4 lines fix a very occasional core dump in chan_skinny found by wedhorn. (issue #13948) ........ 2008-11-21 23:45 +0000 [r158607] Steve Murphy * /, main/features.c: Merged revisions 158606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158606 | murf | 2008-11-21 16:40:46 -0700 (Fri, 21 Nov 2008) | 19 lines Merged revisions 158603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | 11 lines In reference to the fix made for 13871, I was merging the fix into 1.6.0 and realized I missed the code in the h-exten block, and didn't catch it because my test case had the h-exten commented out. So, this corrects the code I missed, as a preventative against another crash report. Tested with the h-exten defined, all is well. ........ ................ 2008-11-21 23:15 +0000 [r158604] Tilghman Lesher * main/pbx.c, /: Merged revisions 158602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ 2008-11-21 22:57 +0000 [r158572] Steve Murphy * /, main/features.c: Merged revisions 158484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | 19 lines Merged revisions 158483 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines (closes issue #13871) Reported by: mdu113 This one is totally my fault. The code doesn't even create a bridge CDR if the channel CDR has POST_DISABLED. I didn't check for that at the end of the bridge. Fixed with a few small insertions. Tested. Looks good. No cdr generated, no crash, no unnecc. data objects created either. ........ ................ 2008-11-21 22:13 +0000 [r158541] Russell Bryant * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 158540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ 2008-11-21 20:43 +0000 [r158450] Kevin P. Fleming * CHANGES, /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt: Merged revisions 158449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files ........ 2008-11-21 19:42 +0000 [r158415] Jason Parker * main/manager.c, /: Merged revisions 158414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158414 | qwell | 2008-11-21 13:40:57 -0600 (Fri, 21 Nov 2008) | 7 lines Make sure we add the Event header for CoreShowChannels. (closes issue #13334) Reported by: srt Patches: 13334_missing_event_header_in_core_show_channel.diff uploaded by srt (license 378) ........ 2008-11-21 17:17 +0000 [r158377] Terry Wilson * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 | twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines Reloading the config and having no changes still initialized some settings to 0. Initialize settings after doing all of the cfg checks. (closes issue #13942) Reported by: davidw Patches: cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: davidw ........ 2008-11-21 01:23 +0000 [r158223-158268] Mark Michelson * /, channels/chan_sip.c: Merged revisions 158265-158266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines Use some magic constants to get the right size for this sscanf statement. Thanks Richard! ........ r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines Use a more expressive constant for a 64-bit scanned int ........ * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines Fix the build for 32-bit systems. %lu is only 32-bits on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. ........ * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines Change the remote user agent session version variable from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj ........ * channels/chan_sip.c: Change this so it actually compiles. Thanks, Terry! 2008-11-20 19:43 +0000 [r158191] Sean Bright * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 | seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 lines Fix one case where the application argument was not converted from a pipe to a comma. This was causing problems with switch statements with empty expressions. (closes issue #13901) Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by seanbright (license 71) Tested by: seanbright Reviewed by: murf ........ 2008-11-20 18:23 +0000 [r158135] Terry Wilson * cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, cdr/cdr_csv.c, cdr/cdr_sqlite3_custom.c, /, cdr/cdr_sqlite.c, cdr/Makefile, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Merged revisions 158072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ 2008-11-20 18:20 +0000 [r158084-158134] Mark Michelson * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c, main/file.c: Merged revisions 158133 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158133 | mmichelson | 2008-11-20 12:20:00 -0600 (Thu, 20 Nov 2008) | 10 lines Merged revisions 158072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ ................ * /, channels/chan_sip.c: Merged revisions 158082 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ 2008-11-20 17:42 +0000 [r158069] Jeff Peeler * /, main/file.c: Merged revisions 158062 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158062 | jpeeler | 2008-11-20 11:37:31 -0600 (Thu, 20 Nov 2008) | 6 lines (closes issue #12929) Reported by: snyfer This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play. ........ 2008-11-20 17:40 +0000 [r158067] Mark Michelson * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ 2008-11-20 00:10 +0000 [r157975] Kevin P. Fleming * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael, channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, main/db1-ast/mpool, res/ais, channels/misdn, res/snmp, Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ ................ 2008-11-19 18:29 +0000 [r157785] Tilghman Lesher * /, configure, configure.ac: Merged revisions 157784 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157784 | tilghman | 2008-11-19 12:28:14 -0600 (Wed, 19 Nov 2008) | 6 lines Add check for t38_terminal_init in spandsp (not found in 0.0.6, so it should fail reasonably) (closes issue #13473) Reported by: genie Patches: 20080916__bug13473.diff.txt uploaded by Corydon76 (license 14) ........ 2008-11-19 13:47 +0000 [r157719-157744] Kevin P. Fleming * /, res/res_agi.c: Merged revisions 157743 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 | kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line correct small bug introduced during API conversion ........ * CHANGES, apps/app_stack.c, include/asterisk/agi.h, /, res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged revisions 157706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ 2008-11-19 01:08 +0000 [r157641] Tilghman Lesher * include/asterisk/logger.h, /, main/logger.c, main/utils.c, include/asterisk/strings.h: Merged revisions 157639 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157639 | tilghman | 2008-11-18 19:02:45 -0600 (Tue, 18 Nov 2008) | 7 lines Starting with a change to ensure that ast_verbose() preserves ABI compatibility in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy in core functions. va_copy() is C99, anyway, and we already require C99 for other purposes, so this isn't really a big change anyway. This change also simplifies some of the core ast_str_* functions. ........ 2008-11-19 01:00 +0000 [r157636] Mark Michelson * /, main/astmm.c: Merged revisions 157632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157632 | mmichelson | 2008-11-18 18:59:48 -0600 (Tue, 18 Nov 2008) | 10 lines If malloc returns NULL, we need to return NULL immediately or else Asterisk will crash when attempting to dereference the NULL pointer (closes issue #13858) Reported by: eliel Patches: astmm.c.patch uploaded by eliel (license 64) ........ 2008-11-19 00:38 +0000 [r157602] Sean Bright * build_tools/make_buildopts_h, makeopts.in, Makefile, /, build_tools/make_version, configure, configure.ac: Merged revisions 157600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 | seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 lines Fix a few build problems on Solaris (and check for an md5 utility in configure instead of the icky loop I was doing before). (closes issue #13842) Reported by: snuffy Patches: bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff uploaded by seanbright (license 71) Tested by: snuffy ........ 2008-11-18 23:59 +0000 [r157429-157596] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 157592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157592 | mmichelson | 2008-11-18 17:59:02 -0600 (Tue, 18 Nov 2008) | 10 lines This change prevents a crash from occurring if res_musiconhold.so is unloaded and then Asterisk is stopped. The problem was that we are not unregistering the ast_moh_destroy function at exit. (closes issue #13761) Reported by: eliel Patches: res_musiconhold.c.patch uploaded by eliel (license 64) ........ * apps/app_voicemail.c, /: Merged revisions 157562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157562 | mmichelson | 2008-11-18 17:28:23 -0600 (Tue, 18 Nov 2008) | 11 lines Fix the logic for when delete=yes when IMAP storage is in use so that the message is deleted from both local and IMAP storage. (closes issue #13642) Reported by: jaroth Patches: deleteyes.patch uploaded by jaroth (license 50) ........ * /, channels/chan_sip.c: Merged revisions 157512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines Merged revisions 157503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ ................ * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines Based on Russell's advice on the asterisk-dev list, I have changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. ........ * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines * Add a lock to be used in the update_call_counter function. * Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) ........ 2008-11-18 19:18 +0000 [r157367] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 157366 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157366 | jpeeler | 2008-11-18 13:16:00 -0600 (Tue, 18 Nov 2008) | 14 lines Merged revisions 157365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines (closes issue #13899) Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. ........ ................ 2008-11-18 18:32 +0000 [r157308] Mark Michelson * apps/app_followme.c, apps/app_dial.c, channels/chan_local.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 157306 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ 2008-11-18 18:20 +0000 [r157304] Steve Murphy * main/config.c, /: Merged revisions 157302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 | murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines (closes issue #13420) Reported by: alex70 Patches: 13420.13539.patch uploaded by murf (license 17) Tested by: murf, awk This fixes two problems: a spurious linefeed insertion probably left over from pre-precomment times. Only generated when category had no previous comments. The other problem: Insertions could get the line-numbering out of whack and generate negative line numbers, causing chunks of line numbers to be emitted, on the scale of the number of lines up to that point in the file. In such cases, abort the looping, and all is well. ........ 2008-11-17 22:39 +0000 [r157255] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 157253 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157253 | tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines Can't use items duplicated off the stack frame in an element returned from a function: in these cases, we have to use the heap, or garbage will result. (closes issue #13898) Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ 2008-11-15 19:49 +0000 [r157108-157166] Kevin P. Fleming * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 157164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line dist-clean should remove dependency information files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice ........ ................ * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines major update to doxygen configuration file: 1) update to doxygen 1.5.x style file, as used in trunk 2) tell doxygen where are header files are, so include-file processing can be done 3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function 4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields 5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit) ........ 2008-11-15 04:30 +0000 [r157040-157042] Russell Bryant * /, channels/chan_sip.c, main/features.c, main/taskprocessor.c: Merged revisions 157041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157041 | russell | 2008-11-14 22:25:57 -0600 (Fri, 14 Nov 2008) | 3 lines Fix a few more places where the case insensitive hash should be used since the comparison is case insensitive. ........ * /, channels/chan_console.c: Merged revisions 157039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157039 | russell | 2008-11-14 22:08:42 -0600 (Fri, 14 Nov 2008) | 3 lines Use the new case insensitive hash function for console interfaces. The comparison function is case insensitive. ........ 2008-11-14 21:21 +0000 [r156963] Mark Michelson * /, channels/chan_sip.c: Merged revisions 156962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156962 | mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 lines Revision 155513 of chan_sip.c in trunk inadvertently removed a very important line to set the "len" field for incoming SIP requests. The result was that all incoming SIP messages appeared to be 0-length, meaning Asterisk could do no meaningful processing of anything SIP-related ........ 2008-11-14 17:04 +0000 [r156913] Tilghman Lesher * main/manager.c, /: Merged revisions 156911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 | tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines Ping is missing the standard double-newline after the event. (closes issue #13903) Reported by: kebl0155 ........ 2008-11-14 16:57 +0000 [r156819-156894] Mark Michelson * apps/app_queue.c, include/asterisk/strings.h: This is the 1.6.1 version of trunk commit 156883. It is functionally equivalent to the 1.6.0 commit * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ ................ 2008-11-14 00:44 +0000 [r156691-156757] Tilghman Lesher * apps/app_while.c, /: Merged revisions 156756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ ................ * main/manager.c, /: Merged revisions 156690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ ................ 2008-11-13 19:29 +0000 [r156654] Brandon Kruse * main/manager.c: Merged revisions 156017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 | pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines Patch by Ryan Brindley -- Make sure that manager refuses any duplicate 'new category' requests in updateconfig (closes issue #13539) ........ 2008-11-13 19:18 +0000 [r156650] Jeff Peeler * main/pbx.c, /: Merged revisions 156649 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156649 | jpeeler | 2008-11-13 13:17:50 -0600 (Thu, 13 Nov 2008) | 6 lines (closes issue #13891) Reported by: smurfix This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null. ........ 2008-11-13 17:12 +0000 [r156614] Mark Michelson * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions 156612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156612 | mmichelson | 2008-11-13 11:07:56 -0600 (Thu, 13 Nov 2008) | 4 lines Kevin sent a note indicating that this change is not necessary, so I am reverting it ........ 2008-11-12 21:36 +0000 [r156389] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) | 12 lines Merged revisions 156386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ ................ 2008-11-12 20:11 +0000 [r156354] Steve Murphy * main/pbx.c, /: Merged revisions 156299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | 26 lines Merged revisions 156297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines It turns out that the 0x0XX00 codes being returned for N, X, and Z are off by one, as per conversation with jsmith on #asterisk-dev; he was teaching a class and disconcerted that this published rule was not being followed, with patterns _NXX, _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should have been. This change, tested on these 3 patterns now picks the proper one. However, this change may surprise users who set up dialplans based on previous behavior, which has been there for what, 2 and half years or so now. ........ ................ 2008-11-12 19:29 +0000 [r156296] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 156295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ ................ 2008-11-12 19:11 +0000 [r156291] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) | 11 lines Merged revisions 156289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ ................ 2008-11-12 19:05 +0000 [r156284-156288] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 156243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600 (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not to be sent, and instead, schedule a task to destroy the iax2 pvt structure 10 seconds later. This allows the IAX2 HANGUP packet to be queued, transmitted, and ACKed before the pvt is destroyed. (closes issue #13645) Reported by: dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14) Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ ........ ................ * apps/app_meetme.c: Fix build (res possibly unused in this function, says gcc) 2008-11-12 18:55 +0000 [r156247] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) | 16 lines Merged revisions 156178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ ................ 2008-11-12 17:48 +0000 [r156171] Mark Michelson * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ ................ 2008-11-12 17:38 +0000 [r156168] Russell Bryant * main/asterisk.c, /: Merged revisions 156166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ ................ 2008-11-12 15:34 +0000 [r156128] Mark Michelson * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions 156127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156127 | mmichelson | 2008-11-12 09:33:11 -0600 (Wed, 12 Nov 2008) | 5 lines Add a couple of AC_SUBST calls to the AST_C_COMPILE_CHECK macro. These missing calls were discovered when working on timerfd support in a separate branch. ........ 2008-11-11 19:52 +0000 [r156005] Tilghman Lesher * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 | tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines Make documentation of update method match documentation and update update2 method to match. Reported by: atis, via -dev mailing list. Fixed by: me ........ 2008-11-10 21:15 +0000 [r155864] Mark Michelson * /, channels/chan_agent.c: Merged revisions 155863 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ ................ 2008-11-10 20:56 +0000 [r155764-155826] Tilghman Lesher * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line I got tired of saying this in every single bugnote referring to this file. ........ * /, main/editline/readline.c: Merged revisions 155763 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155763 | tilghman | 2008-11-10 12:04:30 -0600 (Mon, 10 Nov 2008) | 6 lines Fix memory leak when MALLOC_DEBUG is enabled. (closes issue #13864) Reported by: eliel Patches: readline.c.patch uploaded by eliel (license 64) ........ 2008-11-09 16:32 +0000 [r155556-155672] Sean Bright * configs/chan_dahdi.conf.sample, /: Merged revisions 155671 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155671 | seanbright | 2008-11-09 11:30:29 -0500 (Sun, 09 Nov 2008) | 1 line Fix this as well. Pointed out by tzafrir. ........ * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 155554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ 2008-11-08 21:48 +0000 [r155515-155517] Russell Bryant * /, channels/chan_sip.c, include/asterisk/strings.h: Merged revisions 155516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155516 | russell | 2008-11-08 15:46:43 -0600 (Sat, 08 Nov 2008) | 3 lines - Check for failure when putting the packet in the ast_str - fix a spelling error in a header file ........ * /, channels/chan_sip.c: Merged revisions 155513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155513 | russell | 2008-11-08 15:34:36 -0600 (Sat, 08 Nov 2008) | 3 lines Remove some code that is basically a no-op. Code above this already ensures that the buffer is terminated. ........ 2008-11-07 23:42 +0000 [r155469] Mark Michelson * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines Set the invite state to INV_CANCELLED in a place that makes more sense. Where it was set before, it was impossible to actually delay sending a CANCEL if we had not yet received a provisional response to an INVITE. (closes issue #13626) Reported by: atis Patches: 13626.patch uploaded by putnopvut (license 60) Tested by: atis ........ 2008-11-07 22:29 +0000 [r155396-155400] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 155399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ * /, funcs/func_odbc.c: Merged revisions 155395 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155395 | tilghman | 2008-11-07 16:03:50 -0600 (Fri, 07 Nov 2008) | 2 lines Two bugs relating to colnames found by Marquis42 on #asterisk-dev ........ 2008-11-07 21:16 +0000 [r155362] Mark Michelson * /, configs/voicemail.conf.sample: Merged revisions 155360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, 07 Nov 2008) | 8 lines Remove one more instance of the sample configuration lying about what's possible. The tz cannot be set in a context like this. It can only be set in the general section or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing this out ........ 2008-11-07 20:19 +0000 [r155325] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 155324 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155324 | tilghman | 2008-11-07 14:13:32 -0600 (Fri, 07 Nov 2008) | 7 lines Send call release with unallocated cause instead of normal call clearing, when invalid extension is called. (closes issue #13408) Reported by: adomjan Patches: chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487) ........ 2008-11-07 15:43 +0000 [r155242-155272] Russell Bryant * /, channels/chan_sip.c: Merged revisions 155264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155264 | russell | 2008-11-07 09:42:04 -0600 (Fri, 07 Nov 2008) | 3 lines Remove a bogus ast_free() that Kevin noticed. This was probably just left over from pre-astobj2ified chan_sip. ........ * /, include/asterisk/astobj2.h: Merged revisions 155244 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155244 | russell | 2008-11-07 09:01:02 -0600 (Fri, 07 Nov 2008) | 4 lines Clarify which part of OBJ_MULTIPLE is not implemented, and under what case it is perfectly fine to use. (Inspired by a question I received about my last commit.) ........ * /, channels/chan_sip.c: Merged revisions 155241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155241 | russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines Fix some code in chan_sip that was intended to unlink multiple objects from a container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would only remove a single object. ........ 2008-11-06 22:49 +0000 [r155117-155122] Kevin P. Fleming * res/ael/ael.flex, /, res/ael/ael_lex.c, utils/extconf.c: Merged revisions 155121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 | kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 lines don't blindly assume that Darwin and Cygwin need GLOB_ABORTED defined; only define it if it is not already defined ........ * configure, configure.ac: ensure that an adequately new version of libpri is in place so that chan_dahdi will compile with PRI support 2008-11-06 19:48 +0000 [r155014] Mark Michelson * /, configs/voicemail.conf.sample: Merged revisions 155012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines The documentation listed the ability to set 'maxmsg' per context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. ........ ................ 2008-11-05 22:02 +0000 [r154920] Sean Bright * include/asterisk.h, /: Merged revisions 154919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 | seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 lines Fix a problem found while building res_snmp. ........ 2008-11-05 22:00 +0000 [r154917] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 154428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154428 | tilghman | 2008-11-04 17:03:00 -0600 (Tue, 04 Nov 2008) | 7 lines Switch to using a thread condition to signal that a child thread is ready for work, rather than a busy wait. (closes issue #13011) Reported by: jpgrayson Patches: chan_iax2_find_idle.patch uploaded by jpgrayson (license 492) ........ 2008-11-05 16:14 +0000 [r154690] Steve Murphy * main/channel.c, /: Merged revisions 154687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154687 | murf | 2008-11-05 09:11:11 -0700 (Wed, 05 Nov 2008) | 9 lines Merged revisions 154685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ ................ 2008-11-04 20:52 +0000 [r154367] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 154366 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ ................ 2008-11-04 19:09 +0000 [r154269] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 154268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ ................ 2008-11-04 19:02 +0000 [r154024-154267] Tilghman Lesher * /, channels/chan_h323.c: Merged revisions 154264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ ................ * apps/app_voicemail.c, /: Recorded merge of revisions 154072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008) | 12 lines Merged revisions 154066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) | 5 lines Attempting to expunge a mailbox when the mailstream is NULL will crash Asterisk. (Closes issue #13829) Reported by: jaroth Patch by: me (modified jaroth's patch) ........ ................ * main/rtp.c, /: Merged revisions 154060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ * /, funcs/func_odbc.c: Recorded merge of revisions 154023 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154023 | tilghman | 2008-11-03 15:01:30 -0600 (Mon, 03 Nov 2008) | 4 lines Should have passed the string pointer, not the ast_str structure. (closes issue #13830) Reported by: Marquis ........ 2008-11-03 00:21 +0000 [r153710-153711] Kevin P. Fleming * include/asterisk/compiler.h, apps/app_stack.c, include/asterisk/agi.h, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revision 153709 from trunk ------------------------------------------------------------------------ r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) | 3 lines instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. ------------------------------------------------------------------------ * channels/chan_iax2.c, res/res_jabber.c, channels/chan_oss.c, utils/stereorize.c, main/channel.c, main/manager.c, res/ael/ael_lex.c, main/file.c, pbx/pbx_dundi.c, formats/format_gsm.c, main/asterisk.c, utils/muted.c, /, formats/format_wav.c, apps/app_authenticate.c, res/res_phoneprov.c, res/res_crypto.c, utils/astman.c, res/res_musiconhold.c, res/res_http_post.c, apps/app_queue.c, res/res_config_sqlite.c, agi/eagi-sphinx-test.c, utils/frame.c, channels/chan_dahdi.c, res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c, res/res_agi.c, main/http.c, main/logger.c, channels/chan_h323.c, apps/app_sms.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c, apps/app_adsiprog.c, apps/app_voicemail.c, apps/app_dial.c, channels/chan_sip.c, apps/app_festival.c, main/db1-ast/hash/hash_page.c, res/ael/ael.y, agi/eagi-test.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c, main/utils.c, utils/astcanary.c, formats/format_wav_gsm.c: import gcc 4.3.2 warning fixes from trunk, with a few changes specific to this branch 2008-11-02 20:07 +0000 [r153363-153653] Russell Bryant * include/asterisk/features.h, /: Merged revisions 153652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r153652 | russell | 2008-11-02 14:06:03 -0600 (Sun, 02 Nov 2008) | 10 lines Merged revisions 153651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008) | 2 lines features.h depends on linkedlists.h, so include it ........ ................ * /, channels/chan_sip.c: Merged revisions 153362 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153362 | russell | 2008-11-01 15:41:38 -0500 (Sat, 01 Nov 2008) | 3 lines Ensure that the sip_pvt properly has its refcount incremented when the scheduler holds a reference to it for session timer processing. ........ 2008-10-31 22:11 +0000 [r153266] Terry Wilson * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 153181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ 2008-10-31 20:10 +0000 [r153225] Mark Michelson * main/dial.c, include/asterisk/dial.h: This commit contains the bug fixes and documentation updates which were committed to trunk in revision 153223. I blocked that commit from 1.6.1 since it also contained a new feature. Note to self: Separate commits so that you don't end up with a situation where part of a commit should be merged but part should be blocked from stable branches. 2008-10-31 16:36 +0000 [r153123] Tilghman Lesher * channels/chan_sip.c: Turn off qualify on uncached realtime peers. (Closes issue #13383) 2008-10-30 21:01 +0000 [r152995] Sean Bright * bootstrap.sh: The -I argument to aclocal needs a space before the include directory name. 2008-10-30 20:36 +0000 [r152924-152974] Tilghman Lesher * channels/chan_h323.c: Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) * channels/chan_local.c: Unlock before returning, when extension doesn't exist. (closes issue #13807) Reported by: eliel Patches: chan_local.c.patch uploaded by eliel (license 64) 2008-10-30 19:41 +0000 [r152878-152921] Russell Bryant * channels/chan_sip.c: Fix the sip_peer reference count with respect to scheduler entries for scheduling peer pokes, and scheduling peer poke expirations. * channels/chan_sip.c: Fix the sip_peer reference count with respect to scheduler entries for registration expirations. * include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF(). The reference count of the object _must_ be increased before creating the scheduler entry. Otherwise, you create a race condition where the reference count may hit zero and the object can disappear out from under you. This could also would have incorrectly decreased the reference count in the case that the scheduler add failed. * channels/chan_sip.c: Modify the documentation of the sip_registry struct - Remove a comment that says that the monitor thread is the only one that ever touches these objects. This is no longer the case with TCP. Also, I would eventually like to get the scheduler in its own thread, so this is just a poor assumption to make. - Note that reference counting of these objects with respect to scheduler entries is not complete. There are some leaked references when deleting scheduler entries. 2008-10-30 16:55 +0000 [r152814] Kevin P. Fleming * main/cdr.c: instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse) 2008-10-30 04:29 +0000 [r152690-152777] Tilghman Lesher * configs/extensions.conf.sample: Set up an example stdexten that preserves the original context and extension in the CDR. (Related to issue #13799) Reported by: davidw * main/pbx.c: Track down and fix annoying lock errors. These would occur when merging hints that resulted from a pattern matched hint during a 'dialplan reload'. 2008-10-29 20:55 +0000 [r152648] Mark Michelson * apps/app_directory.c: If there was no named defined in a voicemail.conf mailbox entry, then app_directory would crash when attempting to read that entry from the file. We now check for the NULL or empty string properly so that there will be no crash. (closes issue #13804) Reported by: bluecrow76 2008-10-29 20:16 +0000 [r152645] Terry Wilson * apps/app_queue.c: Small modification to putnopvut's patch to fix this issue. Thanks for all the help, putnopvut! (closes issue #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch uploaded by otherwiseguy (license 396) Tested by: otherwiseguy 2008-10-29 05:52 +0000 [r152606] Steve Murphy * apps/app_queue.c, configs/features.conf.sample, apps/app_dial.c: A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... 2008-10-29 05:35 +0000 [r152573] Russell Bryant * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) 2008-10-29 05:09 +0000 [r152537] Steve Murphy * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c, main/features.c, include/asterisk/pbx.h: The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. 2008-10-28 22:35 +0000 [r152370-152471] Tilghman Lesher * apps/app_voicemail.c: Quoting in the wrong direction (Fixes AST-107) * channels/chan_mgcp.c: Only re-add the io port if it was closed, otherwise reload causes a memory leak. (closes issue #13785) Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64) * apps/app_dial.c: Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy 2008-10-27 23:32 +0000 [r152288] Jeff Peeler * channels/chan_dahdi.c: Buffer policy setting for half is not needed. 2008-10-27 21:53 +0000 [r152173-152217] Tilghman Lesher * channels/chan_local.c: Inherit ALL elements of CallerID across a local channel. (closes issue #13368) Reported by: Peter Schlaile Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 (license 14) * apps/app_stack.c: Oops, only delete the ARG variables once upon release. The following section would have removed them again (removing variables from 2 stack frames, instead of just one). 2008-10-27 16:06 +0000 [r152133] Jason Parker * apps/app_transfer.c: Remove options argument parsing/syntax (it isn't used any longer) (closes issue #13789) Reported by: IgorG Patches: app_transfer.c.diff uploaded by IgorG (license 20) 2008-10-26 20:27 +0000 [r152068] Sean Bright * funcs/func_strings.c: Since passing \0 as the second argument to strchr is valid (and will match the trailing \0 of a string) we need to check that first, otherwise we end up with incorrect results. Fix suggested by reporter. (closes issue #13787) Reported by: meitinger 2008-10-25 11:11 +0000 [r151907] Russell Bryant * main/asterisk.c: Move AMI initialization to occur after loading modules. This prevents a deadlock when someone tries to initiate a module reload from the AMI just as Asterisk is starting. (closes issue #13778) Reported by: hotsblanc Fix suggested by hotsblanc 2008-10-22 20:08 +0000 [r151603] Tilghman Lesher * contrib/scripts/live_ast: Add a contributed script for running Asterisk without installing it, first. (closes issue #11680) Reported by: tzafrir Patches: live_ast_6 uploaded by tzafrir (license 46) 2008-10-22 20:05 +0000 [r151421-151602] Mark Michelson * channels/chan_dahdi.c: Change some logical ands to bitwise ands and add messages alerting that a channel is being ignored if the PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue #13759) Reported by: smurfix Patches: dahdi.patch uploaded by smurfix (license 547) * channels/chan_sip.c: The logic of a strncasecmp call was reversed. (closes issue #13706) Reported by: andrew53 Patches: sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519) * channels/chan_sip.c: Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) * channels/chan_sip.c: Get this compiling in dev-mode * channels/chan_sip.c: If a peer uses any transport other than UDP, then MWI will fail for that peer since sip_alloc will allocate a sip_pvt with a default transport of UDP. This change resets the socket type immediately after allocating the sip_pvt in sip_send_mwi_from_peer, so that the proceeding call to create_addr_from_peer does not fail right away. The socket data from the peer is properly copied to the sip_pvt in create_addr_from_peer. (closes issue #13710) Reported by: andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53 (license 519) * channels/chan_sip.c: When attempting to resolve hostnames, we need to be sure to remove any parameters from the string so that name resolution succeeds. (closes issue #13727) Reported by: fnordian Patches: resolvewithouturiparameter.patch uploaded by fnordian (license 110) 2008-10-21 15:21 +0000 [r151372] Tilghman Lesher * apps/app_mixmonitor.c: Default file modes should always be full read and write, to allow the system administrator to make the decision of what permissions will actually be given, through the use of the process umask. (Closes issue# 13751) 2008-10-21 11:03 +0000 [r151328] BJ Weschke * channels/chan_sip.c: Fix configuration parsing so type=friend still identifies "friend" as a peer even though it is now a legacy configuration verb. (closes issue #13705) reported by: blitzrage patched by: bweschke 2008-10-20 05:06 +0000 [r151135-151245] Kevin P. Fleming * autoconf (added), autoconf/ast_check_pwlib.m4, autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, autoconf/ast_gcc_attribute.m4, bootstrap.sh, autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4, autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4, autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4: break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches. Rename this macro to properly reflect what it does * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the TCP/TLS socket API: 1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address 2008-10-18 02:29 +0000 [r150829] BJ Weschke * main/manager.c: Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke 2008-10-17 17:10 +0000 [r150606-150636] Tilghman Lesher * channels/chan_iax2.c: Make helper call a little safer (suggested by Russell on IRC) * channels/chan_iax2.c, include/asterisk/sched.h: Fix the FRACK! warnings in chan_iax2 when POKE/LAGRQ packets are not answered. 2008-10-16 23:41 +0000 [r150208-150306] Mark Michelson * main/manager.c: Reverting changes from commits 150298 and 150301 since I was mistakenly under the assumption that dialplan functions *always* required that a channel be present. I need to go home earlier, I think :) * main/manager.c: Don't try to call a dialplan function's read callback from the manager's GetVar handler if an invalid channel has been specified. Several dialplan functions, including CHANNEL and SIP_HEADER, do not check for NULL-ness of the channel being passed in. (closes issue #13715) Reported by: makoto And don't forget to return on the error condition * apps/app_sms.c: Answer the channel prior to checking for the 'a' option in app_sms. (closes issue #13675) Reported by: alecdavis Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis (license 585) * configure, configure.ac: Change configure script to search for openais in both /usr/lib and /usr/lib64 since some distros place 64-bit libraries only in the /usr/lib64 directory. (closes issue #13721) Reported by: jcollie Patches: 0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie (license 412) * channels/chan_sip.c: INVITES with proxy auth were sent with a different branch than what was in the invite_branch of a sip_pvt, meaning that if a CANCEL were sent later, the branch in the CANCEL would not match the branch in the latest INVITE sent out, leading to some endpoints responding to the CANCEL with a 481. (closes issue #13714) Reported by: fnordian Patches: invite_branch.patch uploaded by fnordian (license 110) 2008-10-16 16:17 +0000 [r150127] Richard Mudgett * channels/chan_misdn.c: Fix memory leak found by customer 2008-10-16 13:32 +0000 [r149919-149995] Kevin P. Fleming * channels/chan_sip.c: return this logic to where it used to be, *after* the dialog->needdestroy flag has been determined to be set; otherwise, we generate these debug messages every time we inspect every active dialog * apps/app_stack.c: building this module depends on res_agi being built as well * res/res_phoneprov.c: inter-module dependencies should be included in the source code, not just in sample config files * res/res_phoneprov.c: correct file name in message 2008-10-15 21:00 +0000 [r149803] Mark Michelson * channels/chan_sip.c: Make the sip_proxy struct reference counted. This is necessary to allow for a sip_pvt to maintain a reference to a sip_peer's outboundproxy even after the peer has been freed. (closes issue #13700) Reported by: fnordian Patches: 13700.patch uploaded by putnopvut (license 60) Tested by: fnordian 2008-10-15 20:22 +0000 [r149758] BJ Weschke * configs/agents.conf.sample: An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c (closes issue #13709) 2008-10-15 19:09 +0000 [r149589-149688] Tilghman Lesher * funcs/func_odbc.c: Permit data fields to contain more than 255 characters. (closes issue #13631) Reported by: seanbright Patches: 20081015__bug13631.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage * funcs/func_odbc.c: Only set buf to blank before the goto. * codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks memory, because it matches a library malloc() with an ast_free (which, of course, doesn't match up with known allocated memory, so the free fails). (closes issue #13702) Reported by: eliel Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64) * apps/app_echo.c: Minor spacing change (closes issue #13697) Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt uploaded by alecdavis (license 585) 2008-10-15 11:32 +0000 [r149512] Kevin P. Fleming * channels/chan_sip.c: fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support 2008-10-14 23:58 +0000 [r149203-149280] Mark Michelson * CHANGES, apps/app_dial.c: When specifying an invalid timeout to Dial, take it to mean that no timeout is desired. (closes issue #13625) Reported by: atis * channels/chan_sip.c: Change this warning to an error message. Suggestion comes from Sean Bright. Thanks Sean! * channels/chan_sip.c: Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet * apps/app_queue.c: Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) 2008-10-14 22:42 +0000 [r149202] Tilghman Lesher * include/asterisk/hashtab.h, main/chanvars.c, main/config.c, main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c, include/asterisk/chanvars.h, include/asterisk/config.h, include/asterisk/strings.h, res/res_indications.c: Add additional memory debugging to several core APIs, and fix several memory leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx 2008-10-14 21:09 +0000 [r149132] Mark Michelson * channels/chan_sip.c: Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut 2008-10-14 20:17 +0000 [r149063] Tilghman Lesher * apps/app_waitforsilence.c: Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me 2008-10-14 19:42 +0000 [r149060] Leif Madsen * doc/manager_1_1.txt: Add missing documentation for SipShowRegistry action and RegistryEntry event. (closes issue #13342) Reported and patch by: Laureano 2008-10-14 18:59 +0000 [r148918-148986] Tilghman Lesher * apps/app_sms.c: App is ignoring 'p' parameter -- initial pause. (closes issue #13617) Reported by: alecdavis Patches: app_sms.13oct.diff.txt uploaded by alecdavis (license 585) * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. 2008-10-14 17:39 +0000 [r148915] Mark Michelson * channels/chan_local.c: Deadlock prevention in chan_local. (closes issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded by putnopvut (license 60) Tested by: tacvbo 2008-10-14 15:18 +0000 [r148869] Tilghman Lesher * apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher (closes issue #13688) Reported by: irroot Patches: app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me 2008-10-14 11:35 +0000 [r148614-148763] Kevin P. Fleming * channels/chan_sip.c: fix some references to the owner of a private structure that may not be present * Makefile: on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly * channels/chan_sip.c: ensure that *all* fields in the req structure are cleared out before reusing it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled * main/translate.c: it would be nice if this message printing code had actually been tested before it was committed... 2008-10-13 17:56 +0000 [r148562] Steve Murphy * main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. 2008-10-13 15:36 +0000 [r148472] Olle Johansson * channels/chan_sip.c: Sending a 403 after a 200 is considered very bad. (found at SIPit) 2008-10-10 21:22 +0000 [r148375-148377] Mark Michelson * channels/chan_sip.c: The logic used when checking a peer got changed subtly in the "kill the user" commit and caused calls relying on the insecure setting to not work properly. I changed for finding a peer back to how it was prior to that commit. (closes issue #13644) Reported by: pj Patches: 13644_trunkv2.patch uploaded by putnopvut (license 60) Tested by: pj * channels/chan_sip.c: Make sure that the inUse and inRinging fields for a sip peer cannot go below zero. This is a regression from 1.4 and so it will be applied to 1.6.0 as well. (closes issue #13668) Reported by: mjc 2008-10-10 16:37 +0000 [r148269] Tilghman Lesher * apps/app_voicemail.c: User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras 2008-10-10 01:33 +0000 [r148240] Sean Bright * res/res_config_sqlite.c, apps/app_voicemail.c, include/asterisk.h, main/tdd.c, main/cryptostub.c: Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson 2008-10-09 23:55 +0000 [r148151-148161] Mark Michelson * main/manager.c: The priority was unnecessary for the manager atxfer, so it has been removed. Furthermore, now we actually use the Context argument passed to set the transfer context and don't error out if no context is specified. This addresses the actual problems outlined in issue 12158. Regarding the other points brought up, regarding the inability to not transfer to extensions which cannot be represented by DTMF, it is not enough of a constraint that it is worth attempting to rework the feature. (closes issue #12158) Reported by: davidw * apps/app_voicemail.c: Read the callerid in the correct order and make sure to read the Urgent flag value from the IMAP headers. (closes issue #13652) Reported by: jaroth Patches: imapheaders.patch uploaded by jaroth (license 50) 2008-10-09 23:27 +0000 [r148128] Tilghman Lesher * configs/res_ldap.conf.sample: Fix example schema (closes issue #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn (license 503) 2008-10-09 23:20 +0000 [r148115] Mark Michelson * main/features.c: (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! 2008-10-09 20:01 +0000 [r148006-148011] Tilghman Lesher * sounds/Makefile, sounds/sounds.xml: Publish MOH files in sln16 format * apps/app_voicemail.c: When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) 2008-10-09 19:28 +0000 [r147957] Jeff Peeler * main/features.c: (closes issue #13139) Reported by: krisk84 Tested by: krisk84 This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur. 2008-10-09 17:54 +0000 [r147901] Michiel van Baak * include/asterisk/endian.h: only include this for OpenBSD. At least FreeBSD is borked when including it (closes issue #13649) Reported by: ys 2008-10-09 17:47 +0000 [r147898] Tilghman Lesher * configs/extensions.conf.sample: Remove "second form" of extensions, as it no longer applies. Also, cleanup the grammar, formatting, and introduce several clarifications to the text. (Closes issue #13654) 2008-10-09 15:06 +0000 [r147811] Steve Murphy * channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c, main/cli.c, channels/chan_usbradio.c, configure, channels/console_gui.c, utils/extconf.c, main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, include/asterisk/autoconfig.h.in, main/translate.c, channels/vcodecs.c, configure.ac, channels/console_video.c: (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in 2008-10-08 22:33 +0000 [r147719] Mark Michelson * apps/app_meetme.c: Some small tweaks regarding realtime conference announcements. (closes issue #13522) Reported by: DEA Patches: meetme-rt-fixes.txt uploaded by DEA (license 3) 2008-10-08 22:27 +0000 [r147692] Kevin P. Fleming * channels/chan_dahdi.c: when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) 2008-10-08 19:09 +0000 [r147593] Tilghman Lesher * apps/app_sms.c: Correct a typo in the help; also, ensure that the date and time are correctly set, if not specified in the message. (Closes issue #13594, closes issue #13595) Reported by: alecdavis Patches: 20081001__bug13595.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis 2008-10-08 15:10 +0000 [r147519] Mark Michelson * apps/app_speech_utils.c: If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) 2008-10-07 16:54 +0000 [r147196] Sean Bright * apps/app_voicemail.c: Make 'imapsecret' an alias to 'imappassword' in voicemail.conf. 2008-10-07 16:05 +0000 [r147147] Jeff Peeler * main/features.c: Explicitly setting these fields to NULL was done because I wasn't sure if they would be NULL otherwise. Since they will be set automatically, removing. 2008-10-07 15:06 +0000 [r147100] Richard Mudgett * funcs/func_callerid.c: Independent change from branch issue8824 that is not part of COLP. (-r142574 rmudgett) 2008-10-07 12:03 +0000 [r147052] Sean Bright * apps/app_dial.c: Make sure to compare the correct number of characters when special-casing our DAHDI operator mode stuff. Technically, it would work fine, as 'DAH' is currently unique amongst our channel technologies, but as Jared points out: <@jsmith> Sure... as long as the technology starts whith DAH.... but it could be DAHDOO! 2008-10-07 00:13 +0000 [r146972] Terry Wilson * channels/chan_sip.c: A blind transfer to the parking thread would cause a segfault because copy_request accesses dst->data w/o being able to tell whether it is proerly initialized 2008-10-06 23:22 +0000 [r146930] Tilghman Lesher * include/asterisk/threadstorage.h: Update documentation; AST_THREADSTORAGE() in trunk only takes a single argument. 2008-10-06 23:08 +0000 [r146876-146924] Jeff Peeler * include/asterisk/features.h, main/features.c, res/res_agi.c: Similar to r143204, masquerade the channel in the case of Park being called from AGI. ........ * include/asterisk/endian.h: Mvanbaak said this was needed to compile on OpenBSD, so put it in the OpenBSD section. * main/features.c: This commit squashes together three commits because the wrong approach was originally used. (One of the commits was only one line.) 1) r143204: The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel. 2) r143270: Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced. 3) r143475: (the one liner) compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread * main/features.c: fix some comment placement * main/features.c: Explicitly set args in park_call_exec NULL so in the case of no options being passed in, there is no garbage attempted to be used. Also, do not set args to unknown value again if there are no options passed in. 2008-10-06 21:53 +0000 [r146874] Michiel van Baak * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD again 2008-10-06 21:32 +0000 [r146715-146838] Tilghman Lesher * channels/chan_iax2.c, funcs/func_callerid.c, apps/app_speech_utils.c, funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c, funcs/func_cdr.c, funcs/func_math.c: Dialplan functions should not actually return 0, unless they have modified the workspace. To signal an error (and no change to the workspace), -1 should be returned instead. (closes issue #13340) Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14) * channels/chan_local.c: Check whether an extension exists in the _call method, rather than the _alloc method, because we need to evaluate the callerid (since that data affects whether an extension exists). (closes issue #13343) Reported by: efutch Patches: 20080915__bug13343.diff.txt uploaded by Corydon76 (license 14) Tested by: efutch 2008-10-06 16:39 +0000 [r146698] Kevin P. Fleming * channels/chan_dahdi.c: ensure that the private structure for pseudo channels is created without 'leaking' configuration data from other configured channels (closes issue #13555) Reported by: jeffg Patches: issue_13555.patch uploaded by kpfleming (license 421) Tested by: jeffg 2008-10-06 00:23 +0000 [r146557] Sean Bright * utils/Makefile: Quote arguments to cp so we can handle spaces in our paths. 2008-10-05 21:24 +0000 [r146451] Jason Parker * channels/chan_sip.c: Fix silly formatting. 2008-10-04 01:57 +0000 [r146314] Sean Bright * configs/sip_notify.conf.sample: Add ability to remotely reboot snom phones. Also cleaned up and reorganized sip_notify.conf.sample a bit as well. Tested snom reboot on snom 360 and verified snom-check-cfg worked as well. (closes issue #13601) Reported by: mjc Tested by: seanbright 2008-10-03 22:42 +0000 [r146243] Jeff Peeler * main/features.c: remove superfluous reference counting operations in manage_parkinglot since ao2_interator_next increments the ref count automatically 2008-10-03 22:13 +0000 [r146200] Sean Bright * main/cli.c: Resolve a subtle bug where we would never successfully be able to get the first item in the CLI entry list. This was preventing '!' from showing up in either 'help' or in tab completion. (closes issue #13578) Reported by: mvanbaak 2008-10-02 19:31 +0000 [r145960-145964] Russell Bryant * CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0 * CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0 2008-10-02 15:30 +0000 [r145781] Sean Bright * configure, configure.ac: This is much cleaner, methinks. 2008-10-02 15:19 +0000 [r145754] Tilghman Lesher * res/res_odbc.c: Some sanity checks that may have led to prior crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup of incorrectly-used constants. 2008-10-01 23:54 +0000 [r145694] Sean Bright * configure, configure.ac: Try a test compile using the GMime library. Some distros install gmime-config in the base package instead of the -devel package. Now we print a notice and disable GMime support instead of bombing during the main compilation. (closes issue #13583) Reported by: arkadia 2008-10-01 22:24 +0000 [r145557-145609] Mark Michelson * main/features.c: Okay, this should really do it now. While I did manage to fix blind transfers with my last commit here, I also caused an unwanted side-effect. That is, only the first priority of the 'h' extension would be executed when a blind transfer occurred instead of all priorities. Essentially, my last commit corrected the return value of ast_bridge_call. However, the implementation still was not 100% correct. Now it is. * main/features.c: if (!(x) == 0) is the same as if (x). * main/features.c: The logic surrounding the return value of ast_spawn_extension within ast_bridge_call was reversed. This problem was observed when a blind transfer placed from the callee channel of a test call failed. While the problem I am solving here is exactly the same as what was reported in issue #13584, the difference is that this fix I am applying is trunk-only. Issue #13584 was reported against the 1.4 branch, and my tests of 1.4's blind transfers appear to work fine. 2008-10-01 Russell Bryant * Asterisk 1.6.0 released. 2008-09-09 Russell Bryant * Asterisk 1.6.0-rc6 released. 2008-09-09 15:44 +0000 [r142065] Russell Bryant * /, main/features.c: Merged revisions 142064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) | 13 lines Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ ................ 2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 lines Fix a memory leak in chan_oss (closes issue #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel (license 64) ........ 2008-09-09 01:49 +0000 [r141950] Russell Bryant * main/channel.c, /: Merged revisions 141949 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) ........ 2008-09-08 21:07 +0000 [r141808] Russell Bryant * main/pbx.c, /: Merged revisions 141807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) | 15 lines Merged revisions 141806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines When doing an async goto, detect if the channel is already in the middle of a masquerade. This can happen when chan_local is trying to optimize itself out. If this happens, fail the async goto instead of bursting into flames. (closes issue #13435) Reported by: geoff2010 ........ ................ 2008-09-08 Russell Bryant * Asterisk 1.6.0-rc5 released. 2008-09-08 20:19 +0000 [r141746] Jason Parker * Makefile, /, redhat (removed): Merged revisions 141745 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | 8 lines Remove RPM package targets from Makefile (and all associated parts). This has never worked in 1.4, and we decided that it makes no sense to be done here. There are many distros out there that already have "proper" spec files that can be (re)used. Closes issue #13113 Closes issue #10950 Closes issue #10952 ........ ................ 2008-09-08 17:14 +0000 [r141683] Sean Bright * /, build_tools/make_buildopts_h: Merged revisions 141682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on various platforms doesn't choke on the special characters (like ^). (closes issue #13417) Reported by: dougm Patches: 13417.make_buildopts_h.patch uploaded by seanbright (license 71) Tested by: dougm ........ 2008-09-06 20:21 +0000 [r141567] Steve Murphy * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines Merged revisions 141565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs. ........ ................ 2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher * /, res/res_agi.c: Merged revisions 141504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) | 12 lines Merged revisions 141503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) | 4 lines Reverting behavior change (AGI should not exit non-zero on SUCCESS) (closes issue #13434) Reported by: francesco_r ........ ................ 2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines Agent's should not try to call a channel's indicate callback if the channel has been hung up. It will likely crash otherwise ABE-1159 ........ ................ 2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy * main/channel.c, /: Merged revisions 141157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ ................ * pbx/ael/ael-test/ref.ael-vtest25 (added), /, pbx/ael/ael-test/ael-vtest25/extensions.ael, pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged revisions 141115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | 78 lines Merged revisions 141094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines (closes issue #13357) Reported by: pj Tested by: murf (closes issue #13416) Reported by: yarns Tested by: murf If you find this message overly verbose, relax, it's probably not meant for you. This message is meant for probably only two people in the whole world: me, or the poor schnook that has to maintain this code because I'm either dead or unavailable at the moment. This fix solves two reports, both having to do with embedding a function call in a ${} construct. It was tricky because the funccall syntax has parenthesis () in it. And up till now, the 'word' token in the flex stuff didn't allow that, because it would tend to steal the LP and RP tokens. To be truthful, the "word" token was the trickiest, most unstable thing in the whole lexer. I was lucky it made this long without complaints. I had to choose every character in the pattern with extreme care, and I knew that someday I'd have to revisit it. Well, the day has come. So, my brilliant idea (and I'm being modest), was to use the surrounding ${} construct to make a state machine and capture everything in it, no matter what it contains. But, I have to now treat the word token like I did with comments, in that I turn the whole thing into a state-machine sort of spec, with new contexts "curlystate", "wordstate", and "brackstate". Wait a minute, "brackstate"? Yes, well, it didn't take very many regression tests to point out if I do this for ${} constructs, I also have to do it with the $[] constructs, too. I had to create a separate pcbstack2 and pcbstack3 be