Right I will use uac_replace_from to update the from domain but the issue is that when the INVITE comes back from the Asterisk server the original FROM domain is no longer anywhere in the INVITE. So I need to save it with the dialog variables before I send it to Asterisk so that when it comes back from Asterisk I recognize the INVITE thanks to the fU and rU and can find the dialog variable I saved.<div>
<br></div><div>The INVITE that comes back from Asterisk is a new dialog with its own CALLID and totag created by Asterisk.</div><div><br></div><div>I'll try just using the uac_replace_from() function and see if that helps before I get complicated with the dialog variables stuff.<br>
<br><div class="gmail_quote">On Thu, Feb 28, 2013 at 11:56 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u>
<div bgcolor="#ffffff" text="#000000">
<tt>No need to do anything by hand :) - see the uac_replace_from()
function from uac module - it will do all replacements to
guarantee a consistency at dialog level.<br>
<br>
Regards,<br>
</tt><div class="im">
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<br></div><div><div class="h5">
On 02/28/2013 06:10 PM, Duane Larson wrote:
<blockquote type="cite">Yeah. I figure with the Dialog module I will need to
save the from domain before I send it to Asterisk and then when
Asterisk sends it back I will have to match the new INVITE dialog
to the original INVITE so that I can grab that from domain. I
don't see this as being hard to implement.
<div>
<br>
</div>
<div>Thanks for looking at this.<br>
<br>
<div class="gmail_quote">On Thu, Feb 28, 2013 at 10:08 AM,
Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</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">
<div bgcolor="#ffffff" text="#000000"> <tt>Well, do not
know much on Asterisk, so cannot comment :). What I
wanted to point out is that we have the option to do it
on opensips in an easy way -> this will make quite
irrelevant what Asterisk can do.<br>
<br>
Regards,<br>
</tt>
<div>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<br>
</div>
<div>
<div> On 02/28/2013 05:56 PM, Duane Larson
wrote:
<blockquote type="cite">I kind of figured this but
just wanted to check since that post about Asterisk
and the From Header was from back in 2007.
<div><br>
</div>
<div>Thanks<br>
<br>
<div class="gmail_quote">On Thu, Feb 28, 2013 at
7:08 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</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">
<div bgcolor="#ffffff" text="#000000"> <tt>Hi
Duane,<br>
<br>
I guess this leaves you with no
alternatives rather than changing the
domain on opensips - it is not something
complex to do and you can use the dialog
support for that to avoid any dependency
from the end-point devices .<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>
<div> <br>
On 02/28/2013 04:50 AM, Duane Larson
wrote: </div>
</div>
<blockquote type="cite">
<div>
<div>
<div>I wanted to see if I could get
this answered on the OpenSIPS
mailing list even though this kind
of has to do with how Asterisk
works. I am hoping someone has run
into this and figured a way to
resolve the issue.</div>
<div><br>
</div>
<div>I have OpenSIPS set up to be a
proxy for a cluster of Asterisk
servers. When a call comes into
OpenSIPS it relays it to an Asterisk
server, Asterisk handles the call
based on what is in the dialplan and
will always send a new INVITE back
to OpenSIPS and then OpenSIPS sends
the INVITE to the callee.</div>
<div><br>
</div>
<div>This works fine but the new
INVITE that Asterisk generates
changes the domain in the FROM
header to be the IP address of the
Asterisk server. I want to make it
so that Asterisk doesn't change the
From domain or else my only other
option is for OpenSIPS to rewrite
the From domain and change it back
to what it should be. I found the
following post from back in 2007 but
I am not sure if anything has been
changed within Asterisk</div>
<div><br>
</div>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-10836" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-10836</a><br clear="all">
<div><br>
</div>
<div>I can't really change the
fromdomain in my sip.conf file on
the Asterisk servers because the
Asterisk servers are a
multitenant/multidomain.</div>
<div><br>
</div>
<div>Any thoughts on this?</div>
</div>
</div>
<pre><fieldset></fieldset>
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