Summary:	ASTERISK-06189: [patch] No audio on bridges from jan 25 2006
Reporter:	paradise (paradise)	Labels:
Date Opened:	2006-01-25 00:27:08.000-0600	Date Closed:	2008-01-15 16:25:14.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) mydiff.txt
Description:	from mentioned date no audio will be transfered on sip calls. changing date of system to older time solves the problem. ****** ADDITIONAL INFORMATION ****** this problem may happen on other ip channels, but i've not tested.
Comments:	By: paradise (paradise) 2006-01-25 00:30:52.000-0600 this may be related to rtp By: mutilator (mutilator) 2006-01-25 01:14:18.000-0600 Tested this issue out and found these results, running SVN-trunk-r8463M if my system date on the asterisk server is set to anytime before Wed Jan 25 00:18:52 EST 2006 audio works fine. system date with a time anytime after that audio isn't there. during a call from sip phone -> zap channel the audio dissapeares exactly when it hits that date/time on the asterisk server. Calling the other direction zap -> sip yields the same results. By: Roy Sigurd Karlsbakk (rkarlsba) 2006-01-25 01:23:06.000-0600 it's not RTP I had this problem on a system running Zap/IAX2 bridging By: Daniele Gallina (gallysoft) 2006-01-25 02:23:35.000-0600 My system: PSTN Gateway -> SIP -> Asterisk 1.2.2 -> SIP -> Phone Today RTP from PSTN to Asterisk works, but don't works if passed to SIP phone. I switched to Asterisk 1.2.1 and there is no problem. By: Dan Hollis (bani) 2006-01-25 02:45:19.000-0600 mydiff.txt fixed 1.2.2 for me, thanks By: mutilator (mutilator) 2006-01-25 02:47:14.000-0600 also confirming mydiff patch fixed the problem By: Olle Johansson (oej) 2006-01-25 02:49:05.000-0600 Fixed in svn 1.2 (rev 8632) and merged into trunk. By: crich (crich) 2006-01-25 02:53:24.000-0600 will this be fixed in 1.2.2 also ? This looks like a major issue. By: Olle Johansson (oej) 2006-01-25 03:04:38.000-0600 You can patch your server with this patch. 1) Download the attached file mydiff.txt 2) run  patch < mydiff.txt within your source code directory 3) Run make and make install By: Olle Johansson (oej) 2006-01-25 03:06:20.000-0600 Keeping this open so people can find it while we wait for a release. By: Olle Johansson (oej) 2006-01-25 04:05:45.000-0600 We fix 1.2.2 by releasing 1.2.3 - we do not change code that is released, we upgrade to a new version. By: Russell Bryant (russell) 2006-01-25 05:51:12.000-0600 I'm working on the tarball right now.  It will be available shortly. By: Digium Subversion (svnbot) 2008-01-15 16:24:31.000-0600 Repository: asterisk Revision: 8633 _U  trunk/ U   trunk/channel.c ------------------------------------------------------------------------ r8633 | oej | 2008-01-15 16:24:31 -0600 (Tue, 15 Jan 2008) | 2 lines Issue ASTERISK-6189 - patch by markster, imported from 1.2 ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8633 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