-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4093/#review13831
-----------------------------------------------------------

Ship it!


Ship It!

- Joshua Colp


On Nov. 17, 2014, 2:51 a.m., Frankie Chin wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4093/
> -----------------------------------------------------------
> 
> (Updated Nov. 17, 2014, 2:51 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24274
>     https://issues.asterisk.org/jira/browse/ASTERISK-24274
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Currently SLIN12, SLIN24, SLIN32, SLIN44, SLIN48, SLIN96 and SLIN192 are 
> found not working with SIP. The following error will be thrown if one of 
> those codecs is used: chan_sip.c:10718 process_sdp: No compatible codecs, not 
> accepting this offer! 
> 
> What I think the issue is that the codec format isn't being included in the 
> SDP media attributes when one of those codecs is used. Please refer to 
> ASTERISK-24274 for more details. This change updates the main/rtp_engine.c 
> and main/frame.c to ensure all these codecs are supported.
> 
> Note: SLIN and SLIN16 are working fine.
> 
> 
> Diffs
> -----
> 
>   /tags/12.4.0/main/rtp_engine.c 425756 
>   /tags/12.4.0/main/frame.c 425756 
> 
> Diff: https://reviewboard.asterisk.org/r/4093/diff/
> 
> 
> Testing
> -------
> 
> Specified SLIN48 codec in sip.conf of two Asterisk servers. Used AMI to 
> originate a call from Server A to Server B and then put Server B into a 
> conference hosted in Server A. The above mentioned error was no longer 
> reported and the conference was working as expected.
> 
> 
> Thanks,
> 
> Frankie Chin
> 
>

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to