<div dir="ltr"><div>Hi,</div><div>I solved using the dialog and the received port.<br></div><div>But is there any other option ?</div><div><br></div><div>Andrei</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Fri, Apr 12, 2013 at 11:10 PM, Andrei Grav <span dir="ltr"><<a href="mailto:andreigrav@gmail.com" target="_blank">andreigrav@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"><div>Hi,<br></div><div><br></div><div>I am facing some strange situation.</div><div>Opensips is listening on multiple ports on a single public IP 193.xx.xx.20 on ports: 5060, 26999, 36999</div><div>Asterisk is on 193.xx.xx.24:5060</div>
<div><br></div><div>Sometimes Opensips respond from 5060 to a 200OK instead received port.</div><div><br></div><div><br></div><div>U 188.xx.xx.173:53929 -> 193.xx.xx.20:26999<br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 193.xx.xx.20:26999;received=193.xx.xx.20;branch=z9hG4bK46cd.df220482.1.<br>
Via: SIP/2.0/UDP 193.xx.xx.24:5060;rport=5060;received=193.xx.xx.24;branch=z9hG4bK218bf88e.<br>Record-Route: <sip:193.xx.xx.20:26999;lr;r2=on;ftag=as61ef6194;did=49d.46da2342>.<br>Record-Route: <sip:193.xx.xx.20;lr;r2=on;ftag=as61ef6194;did=49d.46da2342>.<br>
Call-ID: <a href="mailto:5ebb1c01580da34c694b320e0627dd7f@sip.mydomain.com" target="_blank">5ebb1c01580da34c694b320e0627dd7f@sip.mydomain.com</a>.<br>From: "User" <<a href="mailto:sip%3A850010@sip.mydomain.com" target="_blank">sip:850010@sip.mydomain.com</a>>;tag=as61ef6194.<br>
To: <<a href="mailto:sip%3A850105@sip.mydomain.com" target="_blank">sip:850105@sip.mydomain.com</a>>;tag=Yab-MAmXkF5lAQq2E6QpifQ8xa9xDoU7.<br>CSeq: 102 INVITE.<br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.<br>
Contact: <sip:850105@188.xx.xx.173:53929;ob>.<br>Supported: replaces, 100rel, timer, norefersub.<br>Content-Type: application/sdp.<br>Content-Length: 294.<br>.<br>v=0.<br>o=- 3574781465 3574781466 IN IP4 188.xx.xx.173.<br>
s=pjmedia.<br>c=IN IP4 188.xx.xx.173.<br>t=0 0.<br>m=audio 4006 RTP/AVP 18 101.<br>c=IN IP4 188.xx.xx.173.<br>a=rtcp:4007 IN IP4 188.xx.xx.173.<br>a=sendrecv.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-15.<br><br><br>U 193.xx.xx.20:5060 -> 188.xx.xx.173:53929<br>ACK sip:850105@188.xx.xx.173:53929;ob SIP/2.0.<br>Via: SIP/2.0/UDP 193.xx.xx.20:26999;branch=z9hG4bK46cd.df220482.3.<br>Via: SIP/2.0/UDP 193.xx.xx.24:5060;rport=5060;received=193.xx.xx.24;branch=z9hG4bK63fbcba6.<br>
Max-Forwards: 69.<br>From: "User" <<a href="mailto:sip%3A850010@sip.mydomain.com" target="_blank">sip:850010@sip.mydomain.com</a>>;tag=as61ef6194.<br>To: <<a href="mailto:sip%3A850105@sip.mydomain.com" target="_blank">sip:850105@sip.mydomain.com</a>>;tag=Yab-MAmXkF5lAQq2E6QpifQ8xa9xDoU7.<br>
Contact: <sip:850010@193.xx.xx.24:5060>.<br>Call-ID: <a href="mailto:5ebb1c01580da34c694b320e0627dd7f@sip.mydomain.com" target="_blank">5ebb1c01580da34c694b320e0627dd7f@sip.mydomain.com</a>.<br>CSeq: 102 ACK.<br>User-Agent: PBX.<br>
Content-Length: 0.<br>.<br></div><div><br></div><div><br></div><div>the last response should send the response from port 26999 to be ok ... or the call is hanged up after 32 seconds</div><div>U 193.xx.xx.20:26999 -> 188.xx.xx.173:53929</div>
<div><br></div><div>any advice ?</div><div><br></div><div>Thank you,</div><div>Andrei</div></div>
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