| Summary: | ASTERISK-16338: Answer not working, maybe chan_oss problem, may be chan_sip problem. | ||
| Reporter: | Massimo Nuvoli (maxnuv) | Labels: | |
| Date Opened: | 2010-07-09 01:34:41 | Date Closed: | 2011-06-07 14:04:41 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_oss |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | This is the way i do the test. from console "dial number@from-sip" the extension number@from-sip make a call trhu SIP provider (tested and working from SIP phone). i answer the "called" phone, i hear nothing (normally i hear something), this is the first problem. i transfer the call to the *199 extension from console "transfer *199@from-sip" the *199 extension is "Answer() MusicOnHold()". I hear nothing. if i use the *198 extension, all things ok the *198 extension is "Answer() PlayBack(silence/1) MusicOnHold()" also if i transfer to another SIP channel no audio, i need to do the "PlayBack(silence/1)" after the Answer() to hear something. This is with asterisk 1.4.32 but also with 1.4.26.2 and some other in the middle. I checked the SIP debug, i see NO difference between the two calls (the one transferred to the *199 and the one transferred to the *198) so this is why i think the problem is chan_oss or console related. I think the "real" problem is the "first" answer, to the SIP channel, but something wrong must be also in the Answer application. The strange is that if i use PlayBack, Voicemail or other apps it works... | ||
| Comments: | By: Stefan Schmidt (schmidts) 2010-07-09 05:11:31 maybe you should check the codec write format on these channels, maybe the first channel has write format slin instead of g711a or what you normally have. it sound like these issue 16287 which depends on a problem with answer. By: Paul Belanger (pabelanger) 2010-07-09 06:29:33 This is a support request, not a bug. Please use the support lists. If you still suspect this is a bug, enable debug logs and upload the appropriate information. --- Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support By: Massimo Nuvoli (maxnuv) 2010-07-12 01:33:01 I am not newbie :-), i now also this is a strange problem. As i made hours of tests to try to isolate the problem and the problem is there this is not a "support" request. :-) Somewere on the asterisk code there is a bug, and i am tryng to find where.. And i made, first, a request on the asterisk-users list... to check if i am wrong or stupid (may be).... but the only response was "sip is working".... that is not my question... Thnks. By: Massimo Nuvoli (maxnuv) 2010-07-12 01:36:27 The SIP channel was first on alaw, then ulaw, then g729, then gsm, and i never found error about the format... (i checked the 16287 issue) After some test i think the real problem is chan_oss "or" the answer and the music_on_hold application, the sip channel i think is ok. By: Paul Belanger (pabelanger) 2010-07-12 06:02:13 As mentioned before, we need a debug log (see below). --- We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt By: Paul Belanger (pabelanger) 2010-07-19 10:26:55 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||