asterisk users - Jun 2004 - Hwo to get CallerID: SIP -> ISDN

Hi!

I trying to configure * in a way, that it uses a different CLIP (Caller-Id
in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far
always the main (1st) number of the number-block is sent to the ISDN.

I have a E100P from Digium and use the zapata stuff (chan_zap).
All SIP calls are coming through an SER.

One idea I had in mind is to assign userid's in SIP, that match the
extension of the number block, e.g. "854". * could then take
the user part of the From header field of the incoming SIP INVITE and
relay this numeric user part (e.g. 854) to the chan_zap, so that the
CLIP in the ISDN appears as the number assigned to SIP user.

Another idea I had was ENUM. But as in ENUM one can only resolve one way,
i.e. E.164-number -> SIP address, *  would have to lookup the whole
number block (every entry) from time to time and cache it in a mapping
table. No so nice solution, I guess.

Does anybody have some experience in this?
Any hints, instructions and HowTo's are warmly welcome.

cheers,
 Bernie

On Fri, 2004-06-18 at 15:16, Bernie Hoeneisen wrote:
> Hi!
> 
> I trying to configure * in a way, that it uses a different CLIP (Caller-Id
> in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far
> always the main (1st) number of the number-block is sent to the ISDN.
> 
> I have a E100P from Digium and use the zapata stuff (chan_zap).
> All SIP calls are coming through an SER.
Have you tried just to use SetCallerID in * before you dispatch the call
to your ZAP channel ?
> One idea I had in mind is to assign userid's in SIP, that match the
> extension of the number block, e.g. "854". * could then take
> the user part of the From header field of the incoming SIP INVITE and
> relay this numeric user part (e.g. 854) to the chan_zap, so that the
> CLIP in the ISDN appears as the number assigned to SIP user.
You can also maintain a database (astdb etc.) which matches the
phoneno.'s against you SIP id's, but your suggestion is easier.
Maintainance free.

It depends a bit on what userbase you have for your SIP users. How much you
manage them or if
they are created/maintained by third party.

Kind regards,
Martin List-Petersen
martin (at) list (dash) petersen (dot) net