Hi all,
I am experiencing some problems while configuring SER in order to fork between different contacts (a SIP UA and a SIP Gateway) and passing through a mediaproxy:
The scenario is the following: (3 calls)
UA1 -> SER : INVITE sip:ramu at xyz.com
SER forks:
SER -> UA2 : INVITE sip:ramu@[local-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {SDP is changed by mediaproxy module
GW -> SER: 183 Session Progress SER -> UA1: 183 Session Progress {SDP is changed by mediaproxy module} Audio session with the mediaproxy is Active GW -> SER: 183 Session Progress SER -> UA1: 183 Session Progress {SDP is changed by mediaproxy module} Audio session with the mediaproxy is Active UA2 -> SER: 180 Ringing SER -> UA1: 180 Ringing
UA2 -> SER: 200 OK SER -> UA1: 200 OK {SDP is changed by mediaproxy module} Audio session with the mediaproxy is Active. SER -> GW: CANCEL ....
The call gets connected, but the problem is that the audio session is mute:
onreply_route[1] {
if (status=~"(183)|2[0-9][0-9]") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); };
};
Can you help me on this ?
Thank you a lot in advance,