[Asterisk-Users] Digium TE405P, caller id and migration to *

[Peter Svensson]

On Mon, 8 Aug 2005, Kib Eki wrote:

> Hi,
> 
> we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our 
> old PBX. So now we could migrate to the * server.
> 
> But, there are two things we can't live with:
> 
> 1. A call from the outside to the old PBX is missing a leading 0 before the number.
> Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as 
> caller number.

See internationalprefix, nationalprefix etc in the file zapata.conf.

> 2. A call made from a SIP client to the outside lacks the extension in the number:
> Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number 
> like 6789-234 when dialing out over the PSTN.

Are you refering to the dialed number or the outgoing caller id (calling 
number)?

Peter