[asterisk-bugs] [JIRA] Commented: (ASTERISK-20357) chan_sip.c: No compatible codecs for this SIP call

Matt Jordan (JIRA) noreply at issues.asterisk.org
Tue Sep 4 11:26:07 CDT 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-20357?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=196558#comment-196558 ] 

Matt Jordan commented on ASTERISK-20357:
----------------------------------------

This was most likely changed by r368218, in which SDPs that include media streams that are not acceptable by Asterisk are rejected, as opposed to Asterisk accepting them.

In general, there were numerous situations in which a UA would present an offer to Asterisk that contained some media that Asterisk could not handle, and there were numerous code paths in which Asterisk would accept the INVITE request, along with whatever media it actually could not process.  This led to some interesting corner cases in which media that had been agreed upon would be sent to Asterisk, and Asterisk could not process it.

I do agree that if a 488 is sent back to the UA, something should indicate to the user on the CLI as to why that occurred.  In this case, the failure of initialize_udptl isn't being propagated to the CLI.  We can at least fix that.

> chan_sip.c: No compatible codecs for this SIP call
> --------------------------------------------------
>
>                 Key: ASTERISK-20357
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20357
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.15.1, 10.6.1, 10.7.0, 10.7.0-digiumphones
>         Environment: Debian 6.0 with kernel 2.6.32-5-amd64 
> sip trunk to a teles iSwitch
>            Reporter: Francesco Usseglio Gaudi
>            Assignee: Matt Jordan
>         Attachments: mydebuglog
>
>
> 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).
> In 1.8.11.1-1digium1~squeeze it works.
> I opened bug ASTERISK-20210 but for my delay it has been closed.

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