Hello,
Can some one let me know the reason that there are these duplicate values in the SDP ( o, c, m, nortpproxy)? Our server was working with kamailio and asterisk on the same machine and had no problem. When we separated kamailio and asterisk on different servers and added the dispatcher module I see this error because of which there is not audio from an incoming PSTN call.
2014/05/15 18:55:19.972332 172.10.30.8:5060 -> 66.136.17.30:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 66.136.17.30:5060 ;rport=5060;branch=z9hG4bK1sansay1225318698rdb6244. Record-Route: sip:54.108.18.75;lr=on;ftag=sansay1225318698rdb6244. Record-Route: sip:sansay1225318698rdb6244@66.136.17.30:5060 ;lr;transport=udp. From: sip:xxxxxx85342@66.136.17.30;tag=sansay1225318698rdb6244. To: sip:12142349395@54.108.18.75;tag=as3414811d. Call-ID: 537507353-0-1838374200@64.136.174.226. CSeq: 1 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uac. Contact: sip:xxxxxxx9395@172.10.30.5:5080. Content-Type: application/sdp. Content-Length: 385. . v=0. o=root 1739301191 1739301191 IN IP4 54.108.18.7554.108.18.7554.108.18.75. s=Asterisk PBX 1.8.17.0. c=IN IP4 54.108.18.7554.108.18.7554.108.18.75. t=0 0. m=audio 388503885038850 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=nortpproxy:yes. a=nortpproxy:yes. a=nortpproxy:yes.
Thank you for the help.
Arun