[Freeswitch-users] Problem in Routing G729A Calls

Brian West brian at freeswitch.org
Wed Nov 5 06:42:59 PST 2008

I would need to see a sip trace of this taking place.  If you're using  
the passthru codec we do pass the fmtp options thru when we receive  


On Nov 5, 2008, at 8:26 AM, shehzad p wrote:

> I have to route the inbound calls of G729A codec.
> Calls comes to my freeswitch with codec G729A (As "annexb=no" is set)
> But when i route calls to termination gateway, calls are dropped  
> (because
> of "annexb=no " is not set)
> Why "annexb=no" is removed while i route the calls?
> How can I set "annexb=no'? (I am using javascript for routing the  
> calls)
> Does following SDP variables can help me in solving above problem?  
> How to
> use those variables?
> http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation
> Warm thanks in advance...
> -- 
> View this message in context: http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.html
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