[OpenSIPS-Users] Issue with From domain coming from Asterisk

[Bogdan-Andrei Iancu]

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 
> <bogdan at opensips.org <mailto:bogdan at 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
>>     <bogdan at opensips.org <mailto:bogdan at 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
>>>         <bogdan at opensips.org <mailto:bogdan at 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?
>>>>
>>>>
>>>>             _______________________________________________
>>>>             Users mailing list
>>>>             Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>>>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>>
>>>         -- 
>>>         --
>>>         *--*--*--*--*--*
>>>         Duane
>>>         *--*--*--*--*--*
>>>         -- 
>>
>>
>>
>>
>>     -- 
>>     --
>>     *--*--*--*--*--*
>>     Duane
>>     *--*--*--*--*--*
>>     -- 
>
>
>
>
> -- 
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> -- 
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