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 href="http://172.25.249.15:5060">172.25.249.15:5060</a> -> <a 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 href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>;tag=as6411602a.</div><div>To: <<a href="http://sip:775008@172.25.249.15:5060">sip:775008@172.25.249.15:5060</a>>;tag=as55ab3180.</div>
<div>Call-ID: <a 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 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 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 href="http://172.25.249.14:5060">172.25.249.14:5060</a> -> <a href="http://172.25.249.15:5060">172.25.249.15:5060</a></div>
<div>ACK <a 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 href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>;tag=as6411602a.</div>
<div>To: <<a href="http://sip:775008@172.25.249.15:5060">sip:775008@172.25.249.15:5060</a>>;tag=as55ab3180.</div><div>Contact: <<a href="mailto:sip%3Aasterisk@172.25.249.14">sip:asterisk@172.25.249.14</a>>.</div>
<div>Call-ID: <a 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 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>