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.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
