Hello Ronald, A 180 Ringing is something that should not have SDP because it's out of band signaling of the exact status of the call, ringing. The PSTN Gateway should return a 183 Session Progress if it wants to deliver inband audio progress. Their SIP implementation doesn't look the best either... so to get it to work you'd either have to hack Asterisk, or get the manufacturer of the PSTN gateway to fix their stuff.
Joshua Colp -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Voermans Sent: Monday, September 26, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Early Media in 100 Ringing 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 [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
