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 them. /b 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... MSP -- View this message in context: http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.