[OpenSIPS-Users] handling DIDs with alias_db

[Bogdan-Andrei Iancu]

Hi Remco,

Try something like that:

     $var(x) = $rU ; # make a backup of original DID (username part of RURI)
     alias_db_lookup() ;  # alias the DID to the main account
     lookup("location");  # see where the main account is registered from
     $rU = $var(x) ; # when sending call to asterisk, for the original 
DID in RURI.

Regards,
Bogdan

On 12/06/2011 01:33 AM, Remco . wrote:
> Hi,
> I'm currently using OpenSIPS as a proxy between PSTN gateways and 
> PBX'es (Asterisk based).
> The PBX'es are registering to OpenSIPS. When a call comes in from the 
> PSTN it get's routed to the PBX correctly.
> However, I would like to use multiple DIDs on the same AOR (e.g. use 
> it as a trunk).
>
> By using the alias_db module, and the||alias_db_lookup() function I'm 
> able to route calls addressed to alias 1001 to user 1000 but the 
> R-URI/contact header gets rewritten and the PBX is not able determine 
> which DID is being called (it comes in with the same R-URI as the user).
>
> What I would like to do is:
> user 1000 at 1.2.3.4 <mailto:1000 at 1.2.3.4>
> Alias 1001 at sip.domain.nl <mailto:1001 at sip.domain.nl> => 
> 1000 at sip.domain.nl <mailto:1000 at sip.domain.nl>
>
> Call comes in, detect whether the R-URI is an alias and point the call 
> to alias at 1.2.3.4 <mailto:alias at 1.2.3.4> where 1.2.3.4 is the IP 
> address listed in the AOR for the user to which the alias is pointing.
> Some posts on the list suggest rewriting the contact using subst() 
> from the textopts module but after some fiddling around with regexes 
> I'm not able to produce the desired result.
>
> Could someone point me in the right direction?
> Thanks in advance,
> Remco.
>
>
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-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"

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