[Freeswitch-users] Problem in Routing G729A Calls
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
> of "annexb=no " is not set)
> Why "annexb=no" is removed while i route the calls?
> Does following SDP variables can help me in solving above problem?
> How to
> use those variables?
> Warm thanks in advance...
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