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

[Kib Eki]

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.

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.

Can anybody tell me how i must change the configuration?

Do you need the zapata.conf?

Thanks in advance and regards