<div dir="ltr">Well ... kamailio is a proxy (not a B2BUA), and in dialog requests should not point at the proxy.<div>If you are paranoid about it, then you can alter signalling by mangling and de-mangling the Contact header for requests and reply to achieve that.</div><div><br></div><div>Regards,</div><div>Ovidiu Sas</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Feb 19, 2015 at 12:59 PM, Andres <span dir="ltr"><<a href="mailto:andres@telesip.net" target="_blank">andres@telesip.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"><span class="">
<div>On 2/18/15 9:44 PM, Will Ferrer wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hi Alex
<div><br>
</div>
<div>Thanks so much for the reply.</div>
<div><br>
</div>
<div>Is there anything that we could do perhaps that is a more
creative solution, for instance not passing the re-invite all
the way to the softphone and just responding from the kamailio
box handling the call?</div>
<div><br>
</div>
<div>We tried this as well actually, but we didn't get it to
work. We just sent a 200 ok from the kamailio box, no sdp or
anything on the packet since we sent it with just send_reply
and the carrier just sent a bye.</div>
<div><br>
</div>
<div>Hopefully there is something clever we could do to correct
the problem, it is preventing us from using alot of our
carriers since the re-invite breaks our clients softphones.</div>
<div><br>
</div>
<div>Thanks again for the assistance.</div>
</div>
</blockquote></span>
We have struggled with this issue ourselves. The problem was that
we did not want our SIP server to behave like an open relay. We
were seeing that the session-timer Re-Invites have a Request-URI
with the IP of the other<br>
endpoint instead of the Proxy. If the SIP server is an open relay
then no problem, but ours is not so the config file was very strict
and dropped the Re-Invite (since the Request-URI had an external IP)
thus dropping the call. The config file could be enhanced by
testing for has_totag() since the Re-Invite has the totag but an
original Invite does not, but the hacker could put a bogus totag and
make calls so its more secure to leave it this way. We ended up
disabling session-timers at some our clients PBXs. Its always a
balancing act between convenience/services and more security. We
chose more security.
<div><div class="h5"><blockquote type="cite">
<div dir="ltr">
<div><br>
</div>
<div>All the best.</div>
<div><br>
</div>
<div>Will Ferrer</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Wed, Feb 18, 2015 at 6:07 PM, Alex
Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="background-color:rgb(255,255,255);line-height:initial">
<div>Kamailio
cannot correct this. This is an endpoint issue. The
whole point of Record-Route is to hairpin sequential
requests (and indeed, their replies) through the proxy.
The endpoints need to comply by affixing the correct
Route header to the end-to-end ACK.</div>
<div><br style="display:initial">
</div>
<div>--<br>
Sent from my BlackBerry. Please excuse errors and brevity.<br>
</div>
<table style="background-color:white;border-spacing:0px" width="100%">
<tbody>
<tr>
<td colspan="2" style="font-size:initial;text-align:initial;background-color:rgb(255,255,255)">
<div>
<div><b>From: </b>Will Ferrer</div>
<div><b>Sent: </b>Wednesday, February 18, 2015
9:01 PM</div>
<div><b>To: </b>Kamailio (SER) - Users Mailing
List</div>
<div><b>Reply To: </b>Kamailio (SER) - Users
Mailing List</div>
<div><b>Subject: </b>[SR-Users] Re-invites from
carrier breaks the call</div>
</div>
</td>
</tr>
</tbody>
</table>
<div>
<div><br>
<div>
<div dir="ltr">Hi All
<div><br>
</div>
<div>We have any issue with re invites coming from
the carrier.</div>
<div><br>
</div>
<div>When a reinvite occurs, our softphone client
gets the invite, sends a 100, and then sends 200
ok. However the 200 ok does not have the
softphones ip in the record route. Since it's
not in the record route the ack from the carrier
never makes it's way all the back to the
softphone.</div>
<div><br>
</div>
<div>This causes the softphone to keep sending 200
oks since it never gets the ack.</div>
<div><br>
</div>
<div>Eventually the softphone gets tired of
sending 200 oks and sends a bye.</div>
<div><br>
</div>
<div>Is there any way that Kamailio can help me
correct for this, or do we need to have our
clients use different softphones? If it has to
be handled via softphones is there even a
softphone that can account for this?</div>
<div><br>
</div>
<div>Thanks for all your assistance in advance.</div>
<div><br>
All the best.</div>
<div><br>
</div>
<div>Will Ferrer</div>
<div><br>
</div>
<div>Switchsoft</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
</div>
</div>
</div>
</div>
<br>
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<br>
</blockquote>
</div>
<br>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
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</pre>
</blockquote>
<br>
<br>
</div></div><span class="HOEnZb"><font color="#888888"><pre cols="72">--
Technical Support
<a href="http://www.cellroute.net" target="_blank">http://www.cellroute.net</a></pre>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature">VoIP Embedded, Inc.<br><a href="http://www.voipembedded.com" target="_blank">http://www.voipembedded.com</a></div>
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