[Asterisk-Users] Voicemail + SIP Message header

[Olle E. Johansson]

Lal, Deepak (Contractor) wrote:

> I am trying to use Asterisk as a "pure" voicemail system and have the following
> setup:
> I have the * setup as a SIP peer to a softswitch. When someone calls a number on
> the softswitch and no one picks up the phone, the softswitch forwards the call
> to the * using SIP. The message header of the SIP INVITE has the number
> originally called in the "To:" field, but the INVITE is still being sent to the
> number asterisk is configured for. 
> 
> Is there any way that I can configure asterisk to "read" the To: field in the
> message header of the SIP INVITE and then go to the mailbox of the corresponding
> number? 
So all INVITES go to the same URI, regardless of the called number?
Is it impossible to change that?

If it is, one could implement a SIPTO variable, but I can't see a general
need for that. Already have a SIPFROM variable in chan_sip2.c (hint,hint).

/Olle