Summary:	ASTERISK-08302: zaptel module (wcte11xp) makes asterisk unable to reproduce audio messages
Reporter:	Caio Begotti (caio1982)	Labels:
Date Opened:	2006-12-07 08:41:21.000-0600	Date Closed:	2008-01-15 16:25:13.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	I was only using ztdummy for some final tests in a 1.4 box and then noticed that after building all the modules I want from the zaptel-1.4, the audio was not being reproduced anymore. Asterisk behavior seems to be okay anyway, but no audio messages can't be heard right now. That's the part of my logs that seems to differ when I'm and when I'm not using the wcte11xp module. I ran "rmmod wcte11xp" then looked again in the same log and both the channel.c messages has vanished and the audio then was played and I heard them fine. [Dec  7 12:30:01] DEBUG[4029] channel.c: Scheduling timer at 160 sample intervals [Dec  7 12:30:01] VERBOSE[4029] logger.c:     -- Playing 'vm-login' (language 'en') [Dec  7 12:30:03] DEBUG[4029] channel.c: Scheduling timer at 0 sample intervals Please let me know what else I can test to give you guys a bit more information.
Comments:	By: Caio Begotti (caio1982) 2006-12-07 08:43:11.000-0600 Sorry, I filled the bug ticket withouth checking its category (applications, general). Could I change it on my own or someone could do so, please? By: Matt O'Gorman (mogorman) 2006-12-07 09:37:11.000-0600 closing this bug as bugs relating to digium hardware should be supported by the support department. (1-800-linux-me).  Secondly are you sure the cards is taking interrupts ? if it is not that would explain lack of audio prompts By: Caio Begotti (caio1982) 2006-12-07 09:49:40.000-0600 I'm sorry Mogorman, but could you explain to me how it's strictly related to and only related to a Digium's hardware fault? I just realized that could be a software/driver problem, since I've never got this kind of situation and Zaptel 1.4 is still a beta code, so that would make some sense after all. The interrupts seems ok to me, so is there anything else I should check before filling a ticket like this next time? PS: I just reopened the bug report to grab more information By: Matt O'Gorman (mogorman) 2006-12-07 09:56:34.000-0600 in this case, where the only difference is the presence of a te110p vs ztdummy it is fairly clear there is some newly introduced timing issue of the real zaptel hw.  As to what it is I am unsure at the moment, however Im sure support could straighten it out for you very quickly, if you have any more questions or concerns you can email me directly, mogorman@digium.com By: Digium Subversion (svnbot) 2008-01-15 16:23:04.000-0600 Repository: asterisk Revision: 8538 U   trunk/channels/chan_sip.c ------------------------------------------------------------------------ r8538 | oej | 2008-01-15 16:23:04 -0600 (Tue, 15 Jan 2008) | 2 lines Importing rev ASTERISK-8302 from 1.2, never send response to ACK (issue ASTERISK-6148) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=8538 By: Digium Subversion (svnbot) 2008-01-15 16:25:13.000-0600 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