my sip traces are below Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 192.168.9.250 port 17722 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.9.251:5060: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.9.250:5060;branch=z9hG4bK111ef687;rport Max-Forwards: 70 From: "pbxserver" <sip:[email protected]>;tag=as66c75bd7 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.13 Date: Wed, 21 Mar 2012 11:19:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 1836379524 1836379524 IN IP4 192.168.9.250 s=Asterisk PBX 1.6.2.13 c=IN IP4 192.168.9.250 t=0 0 m=audio 17722 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 6fxogateway/0722490994 [0K <--- SIP read from UDP:192.168.9.251:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: <sip:[email protected]> From: "pbxserver" <sip:[email protected]>;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: [email protected] Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [0K <--- SIP read from UDP:192.168.9.251:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: <sip:[email protected]>;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: "pbxserver" <sip:[email protected]>;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: [email protected] Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.9.251:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.9.250:5060;branch=z9hG4bK111ef687;rport Max-Forwards: 70 From: "pbxserver" <sip:[email protected]>;tag=as66c75bd7 To: <sip:[email protected]>;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.13 Content-Length: 0 -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
