<------------> Dial (.......) in new stack
== Using SIP RTP CoS mark 5 Audio is at 19614 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:101@sip1.domain.com.ua SIP/2.0 Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299 Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Contact: sip:101@2.2.2.101:5080 Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 06 Aug 2013 10:18:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 319
v=0 o=root 1885227245 1885227245 IN IP4 2.2.2.101 s=Asterisk c=IN IP4 2.2.2.101 t=0 0 m=audio 19614 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
--- -- Called SIP/101@sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0
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<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299 Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV CSeq: 102 INVITE Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0
<-------------> --- (11 headers 0 lines) --- list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes -- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0
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