<div dir="ltr"><div>Hi again,</div><div><br></div>Still Missing 200OK for this call. It'll be helpful to send a complete trace for the call coming in to the Asterisk at first place and then Dialing out to the B-leg whose trace which you've just shared.<div>
<br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <span dir="ltr"><<a href="mailto:blessendor@gmail.com" target="_blank">blessendor@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><br><------------><br> Dial (.......) in new stack<br> <br> <br> == Using SIP RTP CoS mark 5<br>
Audio is at 19614<br>Adding codec 100003 (ulaw) to SDP<br>Adding codec 100002 (gsm) to SDP<br>Adding codec 100004 (alaw) to SDP<br>
Adding codec 100017 (testlaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<div class="im"><br>Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br></div>INVITE <a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a> SIP/2.0<br>
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299<br>Max-Forwards: 70<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1b8070ba<br>
To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>
Contact: <<a href="http://sip:101@2.2.2.101:5080" target="_blank">sip:101@2.2.2.101:5080</a>><br>Call-ID: <a href="mailto:2ac37537499c919f01683582349522d6@sip1.domain.com.ua" target="_blank">2ac37537499c919f01683582349522d6@sip1.domain.com.ua</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk<br>Date: Tue, 06 Aug 2013 10:18:03 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 319<br><br>v=0<br>o=root 1885227245 1885227245 IN IP4 2.2.2.101<br>s=Asterisk<br>c=IN IP4 2.2.2.101<br>t=0 0<br>m=audio 19614 RTP/AVP 0 3 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br> -- Called SIP/<a href="mailto:101@sip1.domain.com.ua" target="_blank">101@sip1.domain.com.ua</a><br>
<br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1bd39f9d<br>Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9<br>CSeq: 10050 INVITE<br>Server: Asterisk<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@2.2.2.101:5080" target="_blank">sip:101@2.2.2.101:5080</a>><br>Content-Length: 0<br><br><br><------------><br>
<br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 100 trying -- your call is important to us<br>Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1b8070ba<br>
To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>Call-ID: <a href="mailto:2ac37537499c919f01683582349522d6@sip1.domain.com.ua" target="_blank">2ac37537499c919f01683582349522d6@sip1.domain.com.ua</a><br>
CSeq: 102 INVITE<br>Server: kamailio (4.0.2 (x86_64/linux))<br>Content-Length: 0<br><br><-------------><br>--- (8 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299<br>Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes><br>Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes><br>Call-ID: <a href="mailto:2ac37537499c919f01683582349522d6@sip1.domain.com.ua" target="_blank">2ac37537499c919f01683582349522d6@sip1.domain.com.ua</a><br>
From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1b8070ba<br>To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV<br>
CSeq: 102 INVITE<br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br>Content-Length: 0<br><br><-------------><br>
--- (11 headers 0 lines) ---<br>list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes><br>list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes><br> -- SIP/sip1.domain.com.ua-0000050f is ringing<br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1bd39f9d<br>Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9<br>CSeq: 10050 INVITE<br>Server: Asterisk<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@2.2.2.101:5080" target="_blank">sip:101@2.2.2.101:5080</a>><br>Content-Length: 0<br><br><br><------------></div>
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<br></blockquote></div><br></div>