[Asterisk-Users] Voicemail + SIP Message header

[Kevin Walsh]

Lal, Deepak (Contractor) [dlal at harris.com] 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? 
> 
It sounds to me as if you're forwarding all VM calls to a single
extension on the Asterisk box, such as 1000, and are then trying to
work out which mailbox the call should be sent to, with no further IDs
to use as a guide.

If you're only using Asterisk as an answering machine (a bit of a
waste, in my view) then you could forward all calls to individual
extensions on the Asterisk box, so extension "2101" on your switch
would defer to "2101 at asteriskbox.example.com" for VM.

Once you have that, you could capture all incoming calls with a single
context in "extensions.conf", such as the following:

    [zzzz]
    exten => _XXXX,1,Answer
    exten => _XXXX,2,Wait(1)
    exten => _XXXX,3,VoiceMail2(su${EXTEN})
    exten => _XXXX,4,Hangup

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