Maybe, on the first pass through opensips you could save the domain part
into a custom hdr (before sending to Asterisk) and configure Asterisk to
propagate this hdr. On the second pass through, opensips will do the
uac_replace_from() based on that hdr...
Just an idea :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/28/2013 09:03 PM, Duane Larson wrote:
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.
The INVITE that comes back from Asterisk is a new dialog with its own
CALLID and totag created by Asterisk.
I'll try just using the uac_replace_from() function and see if that
helps before I get complicated with the dialog variables stuff.
On Thu, Feb 28, 2013 at 11:56 AM, Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
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.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/28/2013 06:10 PM, Duane Larson wrote:
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.
Thanks for looking at this.
On Thu, Feb 28, 2013 at 10:08 AM, Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
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.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/28/2013 05:56 PM, Duane Larson wrote:
I kind of figured this but just wanted to check since that
post about Asterisk and the From Header was from back in 2007.
Thanks
On Thu, Feb 28, 2013 at 7:08 AM, Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
Hi Duane,
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 .
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/28/2013 04:50 AM, Duane Larson wrote:
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.
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.
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
https://issues.asterisk.org/jira/browse/ASTERISK-10836
I can't really change the fromdomain in my sip.conf
file on the Asterisk servers because the Asterisk
servers are a multitenant/multidomain.
Any thoughts on this?
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