Re: [Asterisk-Users] PSTN to SIP gateway

2005-07-15 Thread Nick Kartsioukas
On Fri, Jul 15, 2005 at 01:28:45AM -0400, Jose Raborg wrote: Do you want to route the calls depending on the caller id? Or Do you want to assign a DID to a SIP? The remote SIP device will route the calls appropriately based on the information sent to them (the *ANI*DNIS* sent as an extension),

RE: [Asterisk-Users] PSTN to SIP gateway

2005-07-15 Thread Florian Overkamp
Hi, -Original Message- So far I've gotten Asterisk to say: -- Extension 'XX' in context 'pstn' from '' does not exist. Rejecting call on channel 0/23, span 1 (where XX is the phone number I dialed) So, that's a start, I guess ;)

Re: [Asterisk-Users] PSTN to SIP gateway

2005-07-15 Thread Nick Kartsioukas
On Fri, Jul 15, 2005 at 08:23:14AM +0200, Florian Overkamp wrote: Try: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) The '.' is a wildcard match of unknown length. With your pattern you only accept extensions of 1 digit long. Perfect! Thank you! Hmm...it appears it's not receiving ANI info

RE: [Asterisk-Users] PSTN to SIP gateway

2005-07-14 Thread Jose Raborg
Nick: Do you want to route the calls depending on the caller id? Or Do you want to assign a DID to a SIP? JR -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Kartsioukas Sent: Friday, July 15, 2005 12:22 AM To: asterisk-users@lists.digium.com