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
<[email protected] <mailto:[email protected]>> 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
<[email protected] <mailto:[email protected]>> 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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