[asterisk-dev] RTP interop with Sonus: hack

Joshua Colp jcolp at digium.com
Thu Jan 8 17:21:43 CST 2009


----- "Kristian Kielhofner" <kristian.kielhofner at gmail.com> wrote:
> 
> Me again...
> 
> I think Sonus is expecting the first RFC 2833 event (start) to have a
> duration of 0 and increment from there.  Asterisk currently starts at
> 160 and increments (by 160).  Sonus starts at 0 and increments by 80.
> 
> I've confirmed that Asterisk is able to receive the events from Sonus
> properly.  Sonus just doesn't like what Asterisk is sending...
> 

Easy enough to test! Find the ast_rtp_senddigit_begin function in main/rtp.c and change rtp->send_duration to start out at 0.

-- 
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org



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