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
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