Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP:
As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? U 10.254.254.1:5060 -> 192.168.0.173:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35. Record-Route: <sip:[EMAIL PROTECTED]:5060>. Record-Route: <sip:[EMAIL PROTECTED]:5060;lr;nat=yes>. From: "0161801019" <sip:[EMAIL PROTECTED]>;tag=as02de1b95. To: <sip:[EMAIL PROTECTED]>;tag=00-04094-52dbe3bc-6cf68a723. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: <sip:212.241.48.70:5060>. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. # U 192.168.0.173:5060 -> 192.168.1.103:5062 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265. From: "411" <sip:[EMAIL PROTECTED]>;tag=f93ee2f65c6906cb. To: <sip:[EMAIL PROTECTED]>;tag=as675f246d. Call-ID: [EMAIL PROTECTED] CSeq: 60590 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:[EMAIL PROTECTED]>. Content-Length: 0. . _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
