| Summary: | ASTERISK-06186: Dialstatus Returns Incorrect Results When Peer Is Unrechable | ||
| Reporter: | damin (damin) | Labels: | |
| Date Opened: | 2006-01-24 17:04:19.000-0600 | Date Closed: | 2008-01-15 16:28:31.000-0600 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19", "nocdial|IAX2/boehnlein@pbx1/1216410XXXX") in new stack -- Executing Dial("IAX2/cubix-19", "IAX2/boehnlein@pbx1/1216410XXXX|30") in new stack -- Called boehnlein@pbx1/1216410XXXX Jan 21 19:16:07 WARNING[1114]: chan_iax2.c:6970 socket_read: Call rejected by 207.166.192.188: No authority found -- Hungup 'IAX2/pbx1-21' == No one is available to answer at this time (1:0/0/0) -- Executing Goto("IAX2/cubix-19", "s-NOANSWER|1") in new stack -- Goto (macro-nocdial,s-NOANSWER,1) -- Executing Hangup("IAX2/cubix-19", "") in new stack -- Hungup 'IAX2/cubix-19' Shouldn't that return CONGESTION instead? I thought that NOANSWER was reserved for calls that reach app_dial's timeout limit? Or am I just missing something simple? ****** ADDITIONAL INFORMATION ****** Here is the relevant extensions.conf logic that I am using [e164] ; Dundi exten => _1NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN}) ; Dispatch First Trunk exten => _1NXXNXXXXXX,2,Macro(nocdial,${TRUNK}/${EXTEN}) exten => _1NXXNXXXXXX,3,ResetCDR ; On Failure, Dispatch Second Trunk exten => _1NXXNXXXXXX,4,Macro(nocdial,${TRUNK2}/${EXTEN}) exten => _1NXXNXXXXXX,5,ResetCDR ; Third time is a charm? exten => _1NXXNXXXXXX,6,Macro(nocdial,${TRUNK3}/${EXTEN}) [macro-nocdial] exten => s,1,Dial(${ARG1},30) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Hangup exten => s-BUSY,1,Busy(15) exten => s-BUSY,2,Hangup exten => s-CONGESTION,1,NoOp exten => s-CHANUNAVAIL,1,NoOp exten => s-.,1,Goto(s-NOANSWER,1) | ||
| Comments: | By: damin (damin) 2006-01-24 17:07:56.000-0600 I'd like to find out if the other channel drivers behave the same way when a peer is unavailable. Per Jon Todd's message: From jtodd@loligo.com Tue Jan 24 19:07:05 2006 Date: Sun, 22 Jan 2006 21:23:16 -0800 I would expect an unavailable IAX peer to return "CHANUNAVAIL", actually. But this really does warrant some more discussion about what return codes are passed back to the dialplan, and why. - What should an improperly formatted peer name return? In other words, an impossible destination or a destination that is not in <channel>.conf and not a fully-qualified user/host pair? - What should be accessible via the dialplan for SIP returncodes? IAX return codes? I would want to do something different for a SIP "404" error than for a repeated "407" error, as an example. I know that we would _like_ to have Asterisk treat all channels the same way, but sometimes that doesn't suit the needs of the particular task at hand. While I am a firm believer in generalizing, it's always nice to be able to get under the hood and actually see what is going on if the dialplan author is sophisticated enough to use those tools. In short, I think that you've identified something that probably warrants a bug, but as usual the question to me at least seems larger than just this item. JT By: Kevin P. Fleming (kpfleming) 2006-01-24 18:51:44.000-0600 Fixed in branch-1.2 and trunk. By: Digium Subversion (svnbot) 2008-01-15 16:24:10.000-0600 Repository: asterisk Revision: 8608 U branches/1.2/apps/app_dial.c ------------------------------------------------------------------------ r8608 | kpfleming | 2008-01-15 16:24:09 -0600 (Tue, 15 Jan 2008) | 2 lines ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8608 By: Digium Subversion (svnbot) 2008-01-15 16:24:11.000-0600 Repository: asterisk Revision: 8609 _U trunk/ U trunk/apps/app_dial.c ------------------------------------------------------------------------ r8609 | kpfleming | 2008-01-15 16:24:10 -0600 (Tue, 15 Jan 2008) | 10 lines Merged revisions 8608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8608 | kpfleming | 2006-01-24 19:50:52 -0600 (Tue, 24 Jan 2006) | 2 lines ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8609 By: Digium Subversion (svnbot) 2008-01-15 16:25:14.000-0600 Repository: asterisk Revision: 8679 _U team/oej/astum/ D team/oej/astum/ChangeLog U team/oej/astum/apps/app_dial.c U team/oej/astum/asterisk.c U team/oej/astum/cdr/cdr_pgsql.c U team/oej/astum/channel.c U team/oej/astum/channels/chan_agent.c U team/oej/astum/channels/chan_features.c U team/oej/astum/channels/chan_iax2.c U team/oej/astum/channels/chan_sip.c U team/oej/astum/configs/sip.conf.sample U team/oej/astum/contrib/scripts/safe_asterisk U team/oej/astum/include/asterisk/channel.h U team/oej/astum/rtp.c U team/oej/astum/utils/astman.c ------------------------------------------------------------------------ r8679 | oej | 2008-01-15 16:25:13 -0600 (Tue, 15 Jan 2008) | 230 lines Merged revisions 8517,8523-8524,8531,8538-8539,8548,8554,8560-8561,8563,8571-8572,8574,8582,8587,8589-8597,8599,8609-8610,8618,8620,8633,8642-8643,8654,8664-8665,8667,8676,8678 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r8517 | oej | 2006-01-24 11:36:45 +0100 (Tue, 24 Jan 2006) | 2 lines Whitespace change, extra <tab> added from my tab storage. ................ r8523 | oej | 2006-01-24 12:42:09 +0100 (Tue, 24 Jan 2006) | 2 lines Declaring conn and result static to avoid collission with realtime driver (issue 6336, pressureman) ................ r8524 | oej | 2006-01-24 12:46:29 +0100 (Tue, 24 Jan 2006) | 3 lines - Adding whitespace that I found unused outside - Adding "if (option_debug)" before outputting to DEBUG channel ................ r8531 | oej | 2006-01-24 13:48:44 +0100 (Tue, 24 Jan 2006) | 2 lines - Report SIP reload in manager (issue 5742 with small changes) ................ r8538 | oej | 2006-01-24 14:21:13 +0100 (Tue, 24 Jan 2006) | 2 lines Importing rev ASTERISK-8302 from 1.2, never send response to ACK (issue ASTERISK-6148) ................ r8539 | oej | 2006-01-24 14:53:45 +0100 (Tue, 24 Jan 2006) | 2 lines Issue ASTERISK-6163, FreeBSD compatibility with compilation of func_odbc.c (reported by nulbyte) ................ r8548 | oej | 2006-01-24 18:47:41 +0100 (Tue, 24 Jan 2006) | 2 lines Reverting change in revision 8539 - fixed wrong problem. Sorry. ................ r8554 | oej | 2006-01-24 19:15:20 +0100 (Tue, 24 Jan 2006) | 2 lines Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug ASTERISK-6026) ................ r8560 | oej | 2006-01-24 20:08:44 +0100 (Tue, 24 Jan 2006) | 2 lines Issue ASTERISK-5935: Match realtime non-dynamic peers by IP. (siacali). ................ r8561 | oej | 2006-01-24 20:19:20 +0100 (Tue, 24 Jan 2006) | 2 lines Issue 6114: Don't hangup on bye/also if there's no channel. (gst) ................ r8563 | oej | 2006-01-24 20:29:32 +0100 (Tue, 24 Jan 2006) | 2 lines Blocking fix from 1.2 from being applied again. ................ r8571 | russell | 2006-01-24 21:20:05 +0100 (Tue, 24 Jan 2006) | 2 lines convert ast_channel list to use linked list macros (issue ASTERISK-6178) ................ r8572 | russell | 2006-01-24 21:27:09 +0100 (Tue, 24 Jan 2006) | 2 lines store the list of 'atexit' functions using linked list macros (issue ASTERISK-6169) ................ r8574 | oej | 2006-01-24 21:41:08 +0100 (Tue, 24 Jan 2006) | 2 lines Don't reset scheduled ID until we actually end the scheduled event. ................ r8582 | mattf | 2006-01-24 22:45:42 +0100 (Tue, 24 Jan 2006) | 2 lines Updates from royk to safe_asterisk (ASTERISK-5069) Thanks! ................ r8587 | mattf | 2006-01-24 23:06:37 +0100 (Tue, 24 Jan 2006) | 2 lines Make sure safe_asterisk retains previous script defaults ................ r8589 | kpfleming | 2006-01-24 23:33:58 +0100 (Tue, 24 Jan 2006) | 1 line ................ r8590 | kpfleming | 2006-01-24 23:34:06 +0100 (Tue, 24 Jan 2006) | 1 line ................ r8591 | kpfleming | 2006-01-24 23:38:17 +0100 (Tue, 24 Jan 2006) | 10 lines Merged revisions 8588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8588 | kpfleming | 2006-01-24 16:32:09 -0600 (Tue, 24 Jan 2006) | 2 lines ensure that channel cannot become zombie after we check but before we try to start indications ........ ................ r8592 | kpfleming | 2006-01-24 23:40:20 +0100 (Tue, 24 Jan 2006) | 1 line ................ r8593 | kpfleming | 2006-01-24 23:40:57 +0100 (Tue, 24 Jan 2006) | 1 line ................ r8594 | kpfleming | 2006-01-24 23:41:45 +0100 (Tue, 24 Jan 2006) | 1 line ................ r8595 | kpfleming | 2006-01-24 23:42:43 +0100 (Tue, 24 Jan 2006) | 10 lines Merged revisions 8173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8173 | russell | 2006-01-17 20:49:21 -0600 (Tue, 17 Jan 2006) | 2 lines remove ChangeLog from the 1.2 branch. It will only be present in the tags. ........ ................ r8596 | kpfleming | 2006-01-24 23:43:30 +0100 (Tue, 24 Jan 2006) | 1 line ................ r8597 | kpfleming | 2006-01-24 23:43:57 +0100 (Tue, 24 Jan 2006) | 2 lines clean up remaining already-merged revisions ................ r8599 | kpfleming | 2006-01-24 23:45:41 +0100 (Tue, 24 Jan 2006) | 2 lines remove extraneous characters from property ................ r8609 | kpfleming | 2006-01-25 02:52:58 +0100 (Wed, 25 Jan 2006) | 10 lines Merged revisions 8608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8608 | kpfleming | 2006-01-24 19:50:52 -0600 (Tue, 24 Jan 2006) | 2 lines ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186) ........ ................ r8610 | kpfleming | 2006-01-25 02:53:15 +0100 (Wed, 25 Jan 2006) | 1 line ................ r8618 | russell | 2006-01-25 06:37:29 +0100 (Wed, 25 Jan 2006) | 3 lines don't leak almost 200 bytes for each new channel and store the active channel list using the linked list macros (issue ASTERISK-6170) ................ r8620 | russell | 2006-01-25 06:39:25 +0100 (Wed, 25 Jan 2006) | 1 line ................ r8633 | oej | 2006-01-25 10:50:28 +0100 (Wed, 25 Jan 2006) | 2 lines Issue ASTERISK-6189 - patch by markster, imported from 1.2 ................ r8642 | oej | 2006-01-25 13:01:07 +0100 (Wed, 25 Jan 2006) | 3 lines From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel for pointing this out. ................ r8643 | oej | 2006-01-25 13:11:30 +0100 (Wed, 25 Jan 2006) | 3 lines - Remove unused option to transmit_state_notify - Allow for expiry=0 in subscription requests that only wants *one* update and that's it. ................ r8654 | kpfleming | 2006-01-25 15:52:43 +0100 (Wed, 25 Jan 2006) | 3 lines don't queue a congestion frame on a channel that will be immediately hung up anyway clean up/organize code block ................ r8664 | russell | 2006-01-25 19:12:55 +0100 (Wed, 25 Jan 2006) | 2 lines store agent_pvt list using linked list macros (issue ASTERISK-6182) ................ r8665 | russell | 2006-01-25 19:24:32 +0100 (Wed, 25 Jan 2006) | 3 lines store feature_pvt list using linked list macros (issue ASTERISK-6190, with additional changes to prevent a memory leak in unload_module) ................ r8667 | russell | 2006-01-25 19:41:12 +0100 (Wed, 25 Jan 2006) | 1 line ................ r8676 | russell | 2006-01-25 20:06:37 +0100 (Wed, 25 Jan 2006) | 2 lines use arg parsing macros in the AGENT dialplan function (issue ASTERISK-6078, with small mods) ................ r8678 | russell | 2006-01-25 20:16:14 +0100 (Wed, 25 Jan 2006) | 11 lines Merged revisions 8677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8677 | russell | 2006-01-25 14:14:43 -0500 (Wed, 25 Jan 2006) | 3 lines don't call ast_update_realtime with uninitialized variables if we get a registration with an expirey of 0 seconds (issue ASTERISK-6016) ........ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8679 By: Digium Subversion (svnbot) 2008-01-15 16:28:25.000-0600 Repository: asterisk Revision: 8891 _U team/oej/managerstuff/ D team/oej/managerstuff/ChangeLog U team/oej/managerstuff/apps/app_dial.c U team/oej/managerstuff/apps/app_festival.c U team/oej/managerstuff/apps/app_meetme.c U team/oej/managerstuff/apps/app_milliwatt.c U team/oej/managerstuff/apps/app_queue.c U team/oej/managerstuff/ast_expr2.c U team/oej/managerstuff/ast_expr2.fl U team/oej/managerstuff/ast_expr2.h U team/oej/managerstuff/ast_expr2.y U team/oej/managerstuff/ast_expr2f.c U team/oej/managerstuff/asterisk.c U team/oej/managerstuff/channel.c U team/oej/managerstuff/channels/chan_features.c U team/oej/managerstuff/channels/chan_iax2.c U team/oej/managerstuff/channels/chan_sip.c U team/oej/managerstuff/channels/chan_zap.c U team/oej/managerstuff/loader.c U team/oej/managerstuff/logger.c U team/oej/managerstuff/pbx.c U team/oej/managerstuff/res/res_features.c U team/oej/managerstuff/utils/astman.c ------------------------------------------------------------------------ r8891 | oej | 2008-01-15 16:28:24 -0600 (Tue, 15 Jan 2008) | 202 lines Merged revisions 8112,8122,8124,8134,8140,8162,8173,8194,8232,8242,8276,8281,8320,8347,8394,8412,8414,8418,8429,8433,8437,8445,8537,8562,8573,8588,8600,8608,8619,8632,8666,8677,8710,8729,8758,8785,8808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8112 | kpfleming | 2006-01-17 00:51:37 +0100 (Tue, 17 Jan 2006) | 2 lines do rlimit check _after_ reading config file, in case 'dumpcore' is specified there ........ r8122 | kpfleming | 2006-01-17 14:11:55 +0100 (Tue, 17 Jan 2006) | 2 lines update CLI copyright notice ........ r8124 | mogorman | 2006-01-17 17:55:30 +0100 (Tue, 17 Jan 2006) | 3 lines Fixed code ordering of logger_init and queue_log_init bug 6263 ........ r8134 | mattf | 2006-01-17 19:29:57 +0100 (Tue, 17 Jan 2006) | 2 lines Backport of fix for ASTERISK-5936 ........ r8140 | mogorman | 2006-01-17 21:10:29 +0100 (Tue, 17 Jan 2006) | 3 lines Stop any generators running on a channel when festival is called as described in 5996 ........ r8162 | mogorman | 2006-01-18 01:47:04 +0100 (Wed, 18 Jan 2006) | 4 lines Changed order of autoload so that pbx_ comes before channels, and in doing so cause bug 6002 to not be an issue ........ r8173 | russell | 2006-01-18 03:49:21 +0100 (Wed, 18 Jan 2006) | 2 lines remove ChangeLog from the 1.2 branch. It will only be present in the tags. ........ r8194 | mogorman | 2006-01-18 22:02:06 +0100 (Wed, 18 Jan 2006) | 3 lines Solves issue with the login proccess in meetme patch from 6136 ........ r8232 | russell | 2006-01-19 05:17:45 +0100 (Thu, 19 Jan 2006) | 3 lines fix a seg fault due to assuming that space gets allocatted on the stack in the same order that we declare the variables (issue ASTERISK-6130) ........ r8242 | russell | 2006-01-19 05:56:48 +0100 (Thu, 19 Jan 2006) | 3 lines fix Message-Account header to use the ip address if the fromdomain isn't set (issue ASTERISK-6118) ........ r8276 | tilghman | 2006-01-19 20:14:37 +0100 (Thu, 19 Jan 2006) | 2 lines Bug 6072 - Memory leaks in the expression parser ........ r8281 | oej | 2006-01-19 20:40:28 +0100 (Thu, 19 Jan 2006) | 2 lines Enable "musicclass" setting for sip peers as per the config sample. ........ r8320 | mogorman | 2006-01-20 02:00:46 +0100 (Fri, 20 Jan 2006) | 3 lines solved problem with delayreject and iax trunking bug 4291 ........ r8347 | russell | 2006-01-20 19:34:42 +0100 (Fri, 20 Jan 2006) | 2 lines fix invalid value of prev_q (issue ASTERISK-6142) ........ r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sat, 21 Jan 2006) | 2 lines Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded ........ r8412 | russell | 2006-01-22 00:17:06 +0100 (Sun, 22 Jan 2006) | 2 lines prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111) ........ r8414 | russell | 2006-01-22 00:43:14 +0100 (Sun, 22 Jan 2006) | 2 lines temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111 ........ r8418 | russell | 2006-01-22 03:05:41 +0100 (Sun, 22 Jan 2006) | 3 lines add a modified fix to prevent writing outside of the provided workspace when calculating a substring (issue ASTERISK-6111) ........ r8429 | tilghman | 2006-01-22 09:52:49 +0100 (Sun, 22 Jan 2006) | 2 lines Bug 6281 - Cannot set more than a single header with SIPAddHeader ........ r8433 | bweschke | 2006-01-22 16:13:41 +0100 (Sun, 22 Jan 2006) | 3 lines Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953 ........ r8437 | russell | 2006-01-22 18:47:13 +0100 (Sun, 22 Jan 2006) | 2 lines fix MixMonitor crash (issue ASTERISK-6161, probably others) ........ r8445 | russell | 2006-01-22 20:03:53 +0100 (Sun, 22 Jan 2006) | 2 lines fix memory leak from not freeing the queue member list when freeing an old queue ........ r8537 | oej | 2006-01-24 14:15:13 +0100 (Tue, 24 Jan 2006) | 2 lines Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp) ........ r8562 | oej | 2006-01-24 20:21:15 +0100 (Tue, 24 Jan 2006) | 2 lines Issue 6114: Don't hangup on BYE/ALSO with no channel. ........ r8573 | mattf | 2006-01-24 21:37:30 +0100 (Tue, 24 Jan 2006) | 2 lines Backport fix for ASTERISK-6071, hangup on polarity reversal ........ r8588 | kpfleming | 2006-01-24 23:32:09 +0100 (Tue, 24 Jan 2006) | 2 lines ensure that channel cannot become zombie after we check but before we try to start indications ........ r8600 | russell | 2006-01-24 23:55:32 +0100 (Tue, 24 Jan 2006) | 2 lines completely arbitrary whitespace change for testing something with svnmerge ... ........ r8608 | kpfleming | 2006-01-25 02:50:52 +0100 (Wed, 25 Jan 2006) | 2 lines ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186) ........ r8619 | russell | 2006-01-25 06:38:36 +0100 (Wed, 25 Jan 2006) | 2 lines don't leak almost 200 bytes for each new channel (issue ASTERISK-6170) ........ r8632 | oej | 2006-01-25 10:46:43 +0100 (Wed, 25 Jan 2006) | 2 lines Issue ASTERISK-6276 - the "timebomb" bug. Patch by Markster over GPRS ........ r8666 | russell | 2006-01-25 19:39:44 +0100 (Wed, 25 Jan 2006) | 2 lines fix memory leak (inspired by issue ASTERISK-6190) ........ r8677 | russell | 2006-01-25 20:14:43 +0100 (Wed, 25 Jan 2006) | 3 lines don't call ast_update_realtime with uninitialized variables if we get a registration with an expirey of 0 seconds (issue ASTERISK-6016) ........ r8710 | oej | 2006-01-26 15:39:36 +0100 (Thu, 26 Jan 2006) | 2 lines Issue 5898: Registrations does not get deleted if there's an active SIP dialog ........ r8729 | russell | 2006-01-26 20:42:35 +0100 (Thu, 26 Jan 2006) | 2 lines fix problem with dtmf on e&m (issue ASTERISK-6203) ........ r8758 | tilghman | 2006-01-27 01:52:12 +0100 (Fri, 27 Jan 2006) | 2 lines Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files ........ r8785 | oej | 2006-01-27 09:02:16 +0100 (Fri, 27 Jan 2006) | 2 lines Issue 6362 - Register without Contact: and Expires: fails (reporter: op) ........ r8808 | oej | 2006-01-28 14:52:15 +0100 (Sat, 28 Jan 2006) | 3 lines Issue 6182 - Don't remove scheduled event until it's really done. (reported by malverian) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8891 By: Digium Subversion (svnbot) 2008-01-15 16:28:31.000-0600 Repository: asterisk Revision: 8894 _U team/oej/moduletest/ U team/oej/moduletest/apps/app_dial.c U team/oej/moduletest/apps/app_queue.c U team/oej/moduletest/ast_expr2.c U team/oej/moduletest/ast_expr2.h U team/oej/moduletest/ast_expr2f.c U team/oej/moduletest/asterisk.c U team/oej/moduletest/channel.c U team/oej/moduletest/channels/chan_features.c U team/oej/moduletest/channels/chan_iax2.c U team/oej/moduletest/channels/chan_sip.c U team/oej/moduletest/channels/chan_zap.c U team/oej/moduletest/pbx.c U team/oej/moduletest/utils/astman.c ------------------------------------------------------------------------ r8894 | oej | 2008-01-15 16:28:30 -0600 (Tue, 15 Jan 2008) | 135 lines Merged revisions 8320,8347,8394,8412,8414,8418,8429,8433,8437,8445,8537,8562,8573,8588,8600,8608,8619,8632,8666,8677,8710,8729,8758,8785,8808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8320 | mogorman | 2006-01-20 02:00:46 +0100 (Fri, 20 Jan 2006) | 3 lines solved problem with delayreject and iax trunking bug 4291 ........ r8347 | russell | 2006-01-20 19:34:42 +0100 (Fri, 20 Jan 2006) | 2 lines fix invalid value of prev_q (issue ASTERISK-6142) ........ r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sat, 21 Jan 2006) | 2 lines Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded ........ r8412 | russell | 2006-01-22 00:17:06 +0100 (Sun, 22 Jan 2006) | 2 lines prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111) ........ r8414 | russell | 2006-01-22 00:43:14 +0100 (Sun, 22 Jan 2006) | 2 lines temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111 ........ r8418 | russell | 2006-01-22 03:05:41 +0100 (Sun, 22 Jan 2006) | 3 lines add a modified fix to prevent writing outside of the provided workspace when calculating a substring (issue ASTERISK-6111) ........ r8429 | tilghman | 2006-01-22 09:52:49 +0100 (Sun, 22 Jan 2006) | 2 lines Bug 6281 - Cannot set more than a single header with SIPAddHeader ........ r8433 | bweschke | 2006-01-22 16:13:41 +0100 (Sun, 22 Jan 2006) | 3 lines Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953 ........ r8437 | russell | 2006-01-22 18:47:13 +0100 (Sun, 22 Jan 2006) | 2 lines fix MixMonitor crash (issue ASTERISK-6161, probably others) ........ r8445 | russell | 2006-01-22 20:03:53 +0100 (Sun, 22 Jan 2006) | 2 lines fix memory leak from not freeing the queue member list when freeing an old queue ........ r8537 | oej | 2006-01-24 14:15:13 +0100 (Tue, 24 Jan 2006) | 2 lines Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp) ........ r8562 | oej | 2006-01-24 20:21:15 +0100 (Tue, 24 Jan 2006) | 2 lines Issue 6114: Don't hangup on BYE/ALSO with no channel. ........ r8573 | mattf | 2006-01-24 21:37:30 +0100 (Tue, 24 Jan 2006) | 2 lines Backport fix for ASTERISK-6071, hangup on polarity reversal ........ r8588 | kpfleming | 2006-01-24 23:32:09 +0100 (Tue, 24 Jan 2006) | 2 lines ensure that channel cannot become zombie after we check but before we try to start indications ........ r8600 | russell | 2006-01-24 23:55:32 +0100 (Tue, 24 Jan 2006) | 2 lines completely arbitrary whitespace change for testing something with svnmerge ... ........ r8608 | kpfleming | 2006-01-25 02:50:52 +0100 (Wed, 25 Jan 2006) | 2 lines ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186) ........ r8619 | russell | 2006-01-25 06:38:36 +0100 (Wed, 25 Jan 2006) | 2 lines don't leak almost 200 bytes for each new channel (issue ASTERISK-6170) ........ r8632 | oej | 2006-01-25 10:46:43 +0100 (Wed, 25 Jan 2006) | 2 lines Issue ASTERISK-6276 - the "timebomb" bug. Patch by Markster over GPRS ........ r8666 | russell | 2006-01-25 19:39:44 +0100 (Wed, 25 Jan 2006) | 2 lines fix memory leak (inspired by issue ASTERISK-6190) ........ r8677 | russell | 2006-01-25 20:14:43 +0100 (Wed, 25 Jan 2006) | 3 lines don't call ast_update_realtime with uninitialized variables if we get a registration with an expirey of 0 seconds (issue ASTERISK-6016) ........ r8710 | oej | 2006-01-26 15:39:36 +0100 (Thu, 26 Jan 2006) | 2 lines Issue 5898: Registrations does not get deleted if there's an active SIP dialog ........ r8729 | russell | 2006-01-26 20:42:35 +0100 (Thu, 26 Jan 2006) | 2 lines fix problem with dtmf on e&m (issue ASTERISK-6203) ........ r8758 | tilghman | 2006-01-27 01:52:12 +0100 (Fri, 27 Jan 2006) | 2 lines Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files ........ r8785 | oej | 2006-01-27 09:02:16 +0100 (Fri, 27 Jan 2006) | 2 lines Issue 6362 - Register without Contact: and Expires: fails (reporter: op) ........ r8808 | oej | 2006-01-28 14:52:15 +0100 (Sat, 28 Jan 2006) | 3 lines Issue 6182 - Don't remove scheduled event until it's really done. (reported by malverian) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8894 | ||