| 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 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | 1.8.15.1 10.6.1 10.7.0 10.7.0-digiumphones | Frequency of Occurrence | Constant |
| Related Issues: | |||
| 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 1.8.11.1digium1~squeeze in to 1.8.15.1 compiled from sources. Using 1.8.15.1 call stops working with "chan_sip.c: No compatible codecs for this SIP call" | ||