In a previous query I asked if someone could shed some light on as to why my endpoints do not receive any audio when the call is redirected to an announcement server. After a lot of testing, I believe I have found the problem. When I initiate a call from my end point the end point advises the callee as to the port the RTP traffic will be present. When the call is handed to my PSTN gateway the Gateway responds with its port for RTP in a 183 Session Progress. When that call fails (due to timeout) we want to send it over to Asterisk where an announcement will be played to the caller--however the caller never hears it. The traces show that Asterisk advertises its RTP on a different port that that of the PSTN Gateway. Some of my endpoints (Cisco IP Phone 7940 and Sipura devices) see and listen for the audio on the new advertised port, however my Arris EMTAs do not, it appears as though they are still "tuned in" to the original audio port advised by the PSTN gateway. My question, is is possible to strip away the "m=audio 22040 RTP/AVP 0 8 18 101." from the SIP message? I would like to strip it away in the event of a 183 from my gateway (that advertises the port), but pass it when the call is actually answered. If it is not possible to strip the RTP port information away from the message, what would be the best way in handling a situation like this. Many thanks in advance. kw _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users