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Hello,<br>
<br>
if the ACK goes through the proxy, then means record routing is
used, but I see no Record-Route in 200 reply and no Route in ACK.
Since there is no Record-Route in 200 ok, the ACK has to be sent to
the contact address from the 200 ok.<br>
<br>
Your config snippet from kamailio shows the part of default config
where record routing is handling -- based on the comments -- since
it no Route, it is dropped.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
On 12/26/11 11:03 PM, Lucas Alvarez wrote:
<blockquote
cite="mid:CAJ1UzXTmc3sFv-KzkE_iAqp_nX9z3kMBU+nQh3TeR1dGAzri2A@mail.gmail.com"
type="cite">I have Kamailio 3.2.0 between two asterisk servers,
after the call set, one of the kamailio send the OK from the
INVITE and the return ACK of that message was discarded. This
makes asterisk hangup the call after 5 secs. It's that right?
<div>
<br>
</div>
<div>OK message: </div>
<div><br>
</div>
<div>
<div>U <a moz-do-not-send="true"
href="http://172.25.249.15:5060">172.25.249.15:5060</a>
-> <a moz-do-not-send="true"
href="http://172.25.249.14:5060">172.25.249.14:5060</a></div>
<div>SIP/2.0 200 OK.</div>
<div>Via: SIP/2.0/UDP
172.25.249.14:5060;branch=z9hG4bK09fc3de6;rport=5060.</div>
<div>From: "asterisk" <<a moz-do-not-send="true"
href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>;tag=as6411602a.</div>
<div>To: <<a moz-do-not-send="true"
href="http://sip:775008@172.25.249.15:5060">sip:775008@172.25.249.15:5060</a>>;tag=as55ab3180.</div>
<div>Call-ID: <a moz-do-not-send="true"
href="mailto:547225391b7828402ecaa03e1dab5a86@172.25.249.14">547225391b7828402ecaa03e1dab5a86@172.25.249.14</a>.</div>
<div>CSeq: 102 INVITE.</div>
<div>Server: Asterisk PBX 1.8.7.1.</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH.</div>
<div>Supported: replaces, timer.</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:775008@172.25.249.15:5080">sip:775008@172.25.249.15:5080</a>>.</div>
<div>Remote-Party-ID: "Eus Test" <<a moz-do-not-send="true"
href="mailto:sip%3A3999@172.25.249.14">sip:3999@172.25.249.14</a>>;party=called;privacy=off;screen=no.</div>
<div>Content-Type: application/sdp.</div>
<div>Content-Length: 285.</div>
<div>.</div>
<div>v=0.</div>
<div>o=root 2045590031 2045590031 IN IP4 172.25.249.15.</div>
<div>s=Asterisk PBX 1.8.7.1.</div>
<div>c=IN IP4 172.25.249.15.</div>
<div>t=0 0.</div>
<div>m=audio 11922 RTP/AVP 0 3 8 101.</div>
<div>a=rtpmap:0 PCMU/8000.</div>
<div>a=rtpmap:3 GSM/8000.</div>
<div>a=rtpmap:8 PCMA/8000.</div>
<div>a=rtpmap:101 telephone-event/8000.</div>
<div>a=fmtp:101 0-16.</div>
<div>a=ptime:20.</div>
<div>a=sendrecv.</div>
<div><br>
</div>
</div>
<div><br>
</div>
<div>Discarded ACK:</div>
<div><br>
</div>
<div>
<div>U <a moz-do-not-send="true"
href="http://172.25.249.14:5060">172.25.249.14:5060</a>
-> <a moz-do-not-send="true"
href="http://172.25.249.15:5060">172.25.249.15:5060</a></div>
<div>ACK <a moz-do-not-send="true"
href="http://sip:775008@172.25.249.15:5080">sip:775008@172.25.249.15:5080</a>
SIP/2.0.</div>
<div>Via: SIP/2.0/UDP
172.25.249.14:5060;branch=z9hG4bK6ea5aff6;rport.</div>
<div>From: "asterisk" <<a moz-do-not-send="true"
href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>;tag=as6411602a.</div>
<div>To: <<a moz-do-not-send="true"
href="http://sip:775008@172.25.249.15:5060">sip:775008@172.25.249.15:5060</a>>;tag=as55ab3180.</div>
<div>Contact: <<a moz-do-not-send="true"
href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>.</div>
<div>Call-ID: <a moz-do-not-send="true"
href="mailto:547225391b7828402ecaa03e1dab5a86@172.25.249.14">547225391b7828402ecaa03e1dab5a86@172.25.249.14</a>.</div>
<div>CSeq: 102 ACK.</div>
<div>User-Agent: Asterisk PBX.</div>
<div>Max-Forwards: 70.</div>
<div>Remote-Party-ID: "asterisk" <<a moz-do-not-send="true"
href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>.</div>
<div>Content-Length: 0.</div>
<div>.</div>
</div>
<div><br>
</div>
<div>Kamailio's configuration where the ACK message it's being
discarded:</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div> if ( is_method("ACK") ) {</div>
<div> if ( t_check_trans() ) {</div>
<div> # no loose-route,
but stateful ACK;</div>
<div> # must be an ACK
after a 487</div>
<div> # or e.g. 404 from
upstream server</div>
<div> t_relay();</div>
<div> exit;</div>
<div> } else {</div>
<div> # ACK without
matching transaction ... ignore and discard</div>
<div> exit;</div>
<div> }</div>
<div> }</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>It would be ok if I relay the ack even if it didn't match any
transaction??</div>
<div>Any help would be appreciated.</div>
<div>Regards, </div>
<div><br>
</div>
<div>Lucas </div>
<div><br>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://linkedin.com/in/miconda">http://linkedin.com/in/miconda</a> -- <a class="moz-txt-link-freetext" href="http://twitter.com/miconda">http://twitter.com/miconda</a></pre>
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