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

[Kib Eki]

Andrew Kohlsmith wrote:
> 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.
Misunderstanding: I need to change the calleridnum because there is missing the 
0 before the 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.
> 
> 
> 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.

> -A.
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