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

[Andrew Kohlsmith]

On Monday 08 August 2005 04:03, Kib Eki wrote:
> 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.

This is absolutely trivial to fix.  Anyone who's been able to put * between a 
PRI and a PBX should be able to figure this out without asking the list.  
It's trivial dialplan stuff.

exten => _X.,1,Dial(Zap/g2/0${EXTEN}) kind of trivial.  You may have to debug 
a little to see where or why the 0's disappearing.

> 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.

Again, trivial dialplan stuff.  Your sip.conf will have the callerid for each 
SIP client and you can append that information to the outgoing CID.

-A.