Bogdan, right now it's being forwarded again to the slee server who sent it, as I'm basically using the configuration provided in <a href="http://www.opensips.org/html/docs/modules/1.6.x/dispatcher.html">http://www.opensips.org/html/docs/modules/1.6.x/dispatcher.html</a><br>
<br><div class="gmail_quote">On Wed, Oct 6, 2010 at 4:16 PM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi David,<br>
<br>
Is the BYE replied or forwarded to whatever destination ?<br>
<br>
probably your record routing is somehow broken. OpenSIPS may misroute the BYE because the invalid route set - posting the 200 OK for INVITE and the BYE will help in investigating this.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
David Santiago wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div><div></div><div class="h5">
Hi all,<br>
<br>
I have a running OpenSIPS installation that I'm using for testing purposes.<br>
<br>
The fact is that I'm forwarding requests from a voip provider to a jain slee server and everything is working fine (INVITEs, ACKs, RTP flow,...), except for the BYEs generated from the server side. They reach the OpenSIPs proxy and are not forwarded to the voip provider in order to finish the call.<br>
<br>
I'm not sure if I have to manually setup a route for this to happen, or if this behaviour is only available by using the B2BUA approach in OpenSIPS.<br>
<br>
<br>
Thanks a lot!<br>
<br>
David<br></div></div>
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</blockquote>
<br>
<br>
-- <br>
Bogdan-Andrei Iancu<br>
OpenSIPS Bootcamp<br>
15 - 19 November 2010, Edison, New Jersey, USA<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
</blockquote></div><br>