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Hello,<br>
<br>
changing the R-URI (sip address in the first line of request) can be
done with varables:<br>
<br>
- $ru - the entire r-uri<br>
- $rd - only the domain part of r-uri<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 14/01/16 23:25, Ryan Mottley wrote:<br>
</div>
<blockquote
cite="mid:CANiEHfVC9vgk5_K64NkbisdM7d4=EgxuXBTw+D9MPVwwEWHUug@mail.gmail.com"
type="cite">
<div dir="ltr">Hi,<br>
<br>
We're running a system with Kamailio running in front of
Asterisk just handling registrations and forwarding everything
else to Asterisk. But we're having an issue during hangup on
incoming calls. If the initiator hangs up, the call completes
successfully. But if one of our phones hangs up, the BYE message
comes back with a 404 "Not Found" and the call doesn't hang up
on the carrier side.<br>
<br>
According to the carrier, it's because the IP in the contact on
our ACK message goes to their audio IP while the header of our
BYE points to their signaling IP. <br>
<br>
ACK sip:[Kamailio Pub
IP]:5060;line=sr--rkpsDAp6YAp6DIpZDZmZeI2ZYI26YIRVDcpsDIpsem*
SIP/2.0<br>
Via: SIP/2.0/UDP <b>[Carrier Signaling IP]</b>;branch=z9hG4bK2236.1402e7b4.2<br>
Via: SIP/2.0/UDP <b>[Carrier Audio IP]</b>;received=<b>[Carrier
Audio IP]</b>;branch=z9hG4bK07a8bccb;rport=5060<br>
Route: <a class="moz-txt-link-rfc2396E" href="sip:[KamailioPubIP];r2=on;lr=on;ftag=as67cef00d;nat=yes"><sip:[Kamailio Pub
IP];r2=on;lr=on;ftag=as67cef00d;nat=yes></a>,<a class="moz-txt-link-rfc2396E" href="sip:10.120.0.1;line=sr--rkpsDthVDIhVDNh6Ogo6eKh6eAQs4LRflC2srQRflC2srGqAl-CAP6rZrZkGDmpGed2APtCvlx1"><sip:10.120.0.1;line=sr--rkpsDthVDIhVDNh6Ogo6eKh6eAQs4LRflC2srQRflC2srGqAl-CAP6rZrZkGDmpGed2APtCvlx1></a><br>
From: "+16014477389" <<a class="moz-txt-link-freetext" href="sip:6014477389@">sip:6014477389@</a><b>[Carrier Audio IP]</b>>;tag=as67cef00d<br>
To: <<a class="moz-txt-link-freetext" href="sip:6016025063@">sip:6016025063@</a><b>[Carrier Signaling IP]</b>>;tag=as643b40ca<br>
Contact: <<a class="moz-txt-link-freetext" href="sip:6014477389@">sip:6014477389@</a><b>[Carrier Audio IP]</b>><br>
Call-ID: 4aaefec90826a2a221f0af9500ad211b@<b>[Carrier Audio IP]</b><br>
CSeq: 102 ACK<br>
User-Agent: packetrino<br>
Max-Forwards: 69<br>
Content-Length: 0
<div><br>
BYE <a class="moz-txt-link-freetext" href="sip:6014477389@">sip:6014477389@</a><b>[Carrier Signaling IP] </b>SIP/2.0<br>
Via: SIP/2.0/UDP [Kamailio Pub
IP];branch=z9hG4bK2236.fca983a45913fb510f97e781a85c7392.0<br>
Via: SIP/2.0/UDP
10.120.0.1;branch=z9hG4bKsr-IqktV1L26BCx0jmwZeI2ZYI26YIRVDcpsDIpsem.-EF8-EtCZYmg6edIMehhA4A.AEzyuiZKfPKo7N-qAcWq6D-rsYc4Zp**<br>
Route: <sip:<b>[Carrier Signaling IP]</b>;lr=on><br>
Max-Forwards: 69<br>
From: <a class="moz-txt-link-rfc2396E" href="sip:6016025063@[KamailioPubIP]"><sip:6016025063@[Kamailio Pub IP]></a>;tag=as643b40ca<br>
To: "+16014477389" <<a class="moz-txt-link-freetext" href="sip:6014477389@">sip:6014477389@</a><b>[Carrier Audio IP]</b>>;tag=as67cef00d<br>
Call-ID: 4aaefec90826a2a221f0af9500ad211b@<b>[Carrier Audio
IP]</b><br>
CSeq: 102 BYE<br>
User-Agent: Asterisk PBX 13.6.0<br>
X-Asterisk-HangupCause: Normal Clearing<br>
X-Asterisk-HangupCauseCode: 16<br>
Content-Length: 0
<div><br>
I'm thinking it's happening because their side isn't
configured correctly to handle traffic coming back from a
proxy, but in the meantime is there a way to rewrite the top
of the BYE header to match the "audio IP" they're requesting
it be sent to?</div>
<div><br>
</div>
<div>Thanks!<br>
<br>
-- <br>
Ryan Mottley, Developer<br>
VOXO, LLC<br>
<a moz-do-not-send="true" href="http://voxo.co">voxo.co</a>
- (601)602-5063</div>
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<br>
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<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://miconda.eu">http://miconda.eu</a></pre>
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