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

[Peter Svensson]

On Mon, 8 Aug 2005, Kib Eki wrote:

> >>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.
> > 
> That is set correctly and works between sip clients. it is only a problem when i 
> try to dial out over zap/g1.

Most likely you and your provider are not in agreement on how the calling 
party number should be encoded (number of digits and which Type Of 
Number / Numbering Plan). The TON/NPI is set with the prilocaldialplan 
option. Make sure you send the expected number of digits. You may have to 
do a SetCallerId() before the dial.

Peter