[asterisk-users] Dynamically setting from domain when calling friends

[Rusty Newton]

Actually SIPFROMDOMAIN was documented here:
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
, but SIPFROMUSER was not. They are now both there! :)

On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton <rnewton at digium.com> wrote:
> On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
> <torbjorn.abrahamsson at gmail.com> wrote:
>> I have a problem where I would like to be able to send an arbitrary SIP
>> domain when sending a call to a registered friend. By default the from
>> domain is set to the IP of the Asterisk server, but I would like to set it
>> to something else.
>> The case is that when a call from a foreign domain comes in to the Asterisk,
>> it will connect it to the callee (but with the domain changed). When the
>> callee wants to make a redial from call history, the domain will not be
>> correct.
>> I could probably do something with the fromdomain setting of the friend, but
>> I would like it to be dynamic, ie not having to update the friend definition
>> every time a different domain is used.
>> I understand that I would need to use outbound proxy in the client to
>> prevent it from dialing the domain directly.
>> Is this possible? Any alternatives?
>
> I'm a little confused about what you want to do, however I'll throw
> some information at you in hopes that it will help out.
>
>  I did a little research and found that you can set the outbound From
> header domain and From header user through two channel variables:
> SIPFROMDOMAIN, SIPFROMUSER
>
> They are sparsely documented, but there is an example in extensions.conf
>
> same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
> same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)
>  ; check if we set the FREENUMDOMAIN global variable in [global]
> same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
>  ;    if we did set it, then we'll use it for our outbound dialing
> domain
>
> It looks like they were added in 1.6.2.X of Asterisk, so if you are
> using 1.8.X or above, you should have them.
>
> On your inbound call, you could use the function SIP_HEADER[1] to
> gather the domain and store it for later use when you want to set it
> on the outbound call. Though I'm not sure how you could tell that the
> call was a redial.
>
> [1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
>
> I'm assuming when your SIP client redials that it calls through
> Asterisk and is not dialing the previously caller directly.
>
> Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
> SIPFROMUSER on the wiki*
>
>
> --
> Rusty Newton
> Digium, Inc. | Community Support Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct: +1 256 428 6200
>
> Check us out at: http://digium.com & http://asterisk.org



-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org