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<tt>Hi Dave,<br>
<br>
<br>
<br>
Regards,<br>
<br>
Assuming you do the record-routing for the initial INVITE, can you
post a SIP capture showing the call flow from the OpenSIPS
perspective (containing both legs).<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 13.01.2016 22:54, Dave Lechasseur
wrote:<br>
</div>
<blockquote
cite="mid:F0AEF689-6737-44E1-9DEF-B885A03A134A@sbktelecom.com"
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<div>Hi Eric,</div>
<div><br>
</div>
<div>Thank you for the quick answer.</div>
<div>With record_route it go thought the proxy but every
in-session messages aren’t sent to the PBX or to the phone
from the PBX.</div>
<div><br>
</div>
<div>Just to make sure I’ve tried adding : xlog("DEBUG : METHOD
$rm”); </div>
<div>right after : route { </div>
<div><br>
</div>
<div>and I don’t pick any log from it.</div>
<div><br>
</div>
<div>Thank you,</div>
<div>Dave L.</div>
<div><br>
</div>
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<span style="font-weight:bold">From: </span><<a
moz-do-not-send="true"
href="mailto:users-bounces@lists.opensips.org"><a class="moz-txt-link-abbreviated" href="mailto:users-bounces@lists.opensips.org">users-bounces@lists.opensips.org</a></a>>
on behalf of Eric Tamme <<a moz-do-not-send="true"
href="mailto:eric@uphreak.com">eric@uphreak.com</a>><br>
<span style="font-weight:bold">Reply-To: </span>OpenSIPS
users mailling list <<a moz-do-not-send="true"
href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br>
<span style="font-weight:bold">Date: </span>Wednesday,
January 13, 2016 at 3:44 PM<br>
<span style="font-weight:bold">To: </span>OpenSIPS users
mailling list <<a moz-do-not-send="true"
href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br>
<span style="font-weight:bold">Subject: </span>Re:
[OpenSIPS-Users] T_Relay Re-Invite<br>
</div>
<div><br>
</div>
<div>
<div bgcolor="#FFFFFF" text="#000000">You need to record route
the initial request so that your proxy stays in the
signalling path. This doesnt have anything to do with the
tm module, but rather building up the routeset and loose
routing.<br>
<br>
-Eric<br>
<br>
<div class="moz-cite-prefix">On 01/13/2016 01:42 PM, Dave
Lechasseur wrote:<br>
</div>
<blockquote
cite="mid:A460AF9E-B62D-4CD3-B0B9-F917C116A68A@sbktelecom.com"
type="cite">
<div>
<div>
<div>Hi everyone,</div>
</div>
</div>
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<div><br>
</div>
<div>I have a problem with t_relay.</div>
<div><br>
</div>
<div>When the session is established (last 200 OK) the
phone receive a in-dialog invite from the PBX
directly and since there every packet don’t go
thought OpenSIPS but goes directly to the PBX
meaning that I have no way to do anything on the
active channel and I don’t receive the BYE message
on OpenSIPS, it go directly between the phone and
the PBX.</div>
<div><br>
</div>
<div>Is there a way to not have this ?</div>
<div><br>
</div>
<div>Thank you for your help !</div>
<div><br>
</div>
<div>Dave L.</div>
</div>
</span><br>
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