|Summary:||ASTERISK-20210: chan_sip.c: No compatible codecs for this SIP call|
|Reporter:||Francesco Usseglio Gaudi (cecco)||Labels:|
|Date Opened:||2012-08-10 04:27:07||Date Closed:||2012-08-29 09:04:26|
|Versions:||126.96.36.199 10.6.1 10.7.0 10.7.0-digiumphones||Frequency of|
|Environment:||Debian 6.0 with kernel 2.6.32-5-amd64||Attachments:||( 0) mydebuglog|
|Description:||Incoming call from a sip channel doesn't work. With asterisk svn, 10.5 and 10.4 and 1.8 with same config and codecs it works without problem. Outgoing call on that sip channel always works (whatever version). |
I have also tried 10.8.0-rc1 but it has the same problem
|Comments:||By: Rusty Newton (rnewton) 2012-08-10 10:52:55.333-0500|
Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelineshttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).
This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!
Please include the mentioned SIP trace, preferably a packettrace gathered with tcpdump or a similar method, plus SIP debug from an Asterisk full log https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: Matt Jordan (mjordan) 2012-08-29 09:04:17.430-0500
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
By: Francesco Usseglio Gaudi (cecco) 2012-09-04 10:11:42.256-0500
debug log as requested
By: Francesco Usseglio Gaudi (cecco) 2012-09-04 10:12:00.516-0500
Sorry for delay. Now I have the log.
In this log, i have use the same config files from 188.8.131.52digium1~squeeze in to 184.108.40.206 compiled from sources.
Using 220.127.116.11 call stops working with "chan_sip.c: No compatible codecs for this SIP call"