Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060 Record-Route: sip:192.168.0.192;lr;ftag=as2b21d944 Call-ID: 58eb00885daef7ff3a67ad0e235e817a@14.98.22.110 From: sip:68XXXXX@10.0.X.X;tag=as2b21d944 To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-Huku2c07186a1 CSeq: 102 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE Contact: sip:09413745250@10.0.X.X :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff User-Agent: ZTE Softswitch/1.0.0 Require: timer Session-Expires: 7200;refresher=uac Content-Length: 182 Content-Type: application/sdp
v=0 o=- 1936 20890 IN IP4 10.0.X.X s=SBC call c=IN IP4 10.0.X.X t=0 0 m=audio 37874 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 3.236.72.101:5060 ;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060 Record-Route: sip:192.168.0.192;lr;ftag=as2b21d944 Call-ID: 58eb00885daef7ff3a67ad0e235e817a@14.98.22.110 From: sip:68XXXXX@10.0.X.X;tag=as2b21d944 To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-Huku2c07186a1 CSeq: 102 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE Contact: sip:09413745250@10.0.X.X :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff User-Agent: ZTE Softswitch/1.0.0 Require: timer Session-Expires: 7200;refresher=uac Content-Length: 182 Content-Type: application/sdp
v=0 o=- 1936 20890 IN IP4 10.0.X.X s=SBC call c=IN IP4 10.0.X.X t=0 0 m=audio 37874 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000/1
Regards Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja kashishraheja1809@gmail.com wrote:
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides.
However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks. Regards Kashish +919413745250