<div>Hi,</div>
<div> </div>
<div>Maybe you can check the contact of INVITE from sip proxy to callee.</div>
<div>Try fix_contact () for INVITE.</div>
<div> </div>
<div> </div>
<div>Regards,</div>
<div>Lesley<br><br> </div>
<div><span class="gmail_quote">2007/6/4, Andreas Granig <<a href="mailto:agranig@sipwise.com">agranig@sipwise.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi,<br><br>I'm going crazy with this problem, so maybe anyone here can see any<br>error or can give a hint how to debug this.
<br><br>I've two clients behind (the same) NAT. INVITE goes thru, so does 100<br>and 180. But the 200 isn't relayed because of the following syslog error:<br><br>./sbin/openser[23933]: ERROR: udp_send:<br>sendto(sock,0x81816a8,814,0,0xb5b7f168,16): Operation not permitted(1)
<br>./sbin/openser[23933]: msg_send: ERROR: udp_send failed<br><br>I can't find any error (the two Via headers are exactly the same), so<br>here is the 180 which is relayed without problems, and below the 200,<br>which causes the error:
<br><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP <a href="http://84.19.176.11">84.19.176.11</a>;branch=z9hG4bK6351.1b30ad43.0<br>Via: SIP/2.0/UDP<br><a href="http://192.168.123.100:2083">192.168.123.100:2083</a>;received=<a href="http://83.65.24.161">
83.65.24.161</a>;branch=z9hG4bK-kq6qyyeowiei;rport=12499<br>From: "Foo Bar" <<a href="mailto:sip:foo1000@linguin.org">sip:foo1000@linguin.org</a>>;tag=xuu6a2tp6b<br>To: <<a href="mailto:sip:2000@linguin.org">
sip:2000@linguin.org</a>;user=phone>;tag=2091138726<br>Call-ID: 3c27e5729c40-csdi3p954zyn@snom360-0004132306CE<br>CSeq: 2 INVITE<br>Server: Cisco ATA 186 v3.1.1 atasip (040629A)<br>Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
<br>Content-Length: 0<br><br><br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://84.19.176.11">84.19.176.11</a>;branch=z9hG4bK6351.1b30ad43.0<br>Via: SIP/2.0/UDP<br><a href="http://192.168.123.100:2083">192.168.123.100:2083
</a>;received=<a href="http://83.65.24.161">83.65.24.161</a>;branch=z9hG4bK-kq6qyyeowiei;rport=12499<br>Record-Route: <sip:<a href="http://84.19.176.11">84.19.176.11</a>;lr=on;ftag=xuu6a2tp6b;nat=yes><br>From: "Foo Bar" <
<a href="mailto:sip:foo1000@linguin.org">sip:foo1000@linguin.org</a>>;tag=xuu6a2tp6b<br>To: <<a href="mailto:sip:2000@linguin.org">sip:2000@linguin.org</a>;user=phone>;tag=2091138726<br>Call-ID: 3c27e5729c40-csdi3p954zyn@snom360-0004132306CE
<br>CSeq: 2 INVITE<br>Contact: <sip:foo2000@192.168.123.103:5060;transport=udp><br>Server: Cisco ATA 186 v3.1.1 atasip (040629A)<br>Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER<br>Supported: replaces
<br>Content-Length: 204<br>Content-Type: application/sdp<br><br>v=0<br>o=foo2000 11828 11828 IN IP4 <a href="http://192.168.123.103">192.168.123.103</a><br>s=ATA186 Call<br>c=IN IP4 <a href="http://192.168.123.103">192.168.123.103
</a><br>t=0 0<br>m=audio 16384 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000/1<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br>Any ideas?<br><br>Regards,<br>Andreas<br><br>PS: this happens with both OpenSER 1.1 and
1.2<br><br><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@openser.org">Users@openser.org</a><br><a href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users
</a><br></blockquote></div><br>