Hi there!
What I'm trying to implement with my cisco router is use it into a FreePBX (Asterisk) PBX as analog gateway for lines and FXS subscriber for analog phones.
All is working great, but I have a little issue with dial.patterns when i dial a call from any FXS.
First issue, when I set Outbound roules at PBX it need to match with the local telco and with the cisco device. So what's better? set a dial-pattern in the router to pass it transparent without any manipulation or set some?
I have only implement 9.... and 8... to identify two different lines because one is a telco line and other is a door telco device.
This works great when I call from a PBX soft phone but when I want to use the FXS phone line the only way to dial successfully a call was use ........ (8) as dial pattern. But if the phone have more than 8 digits it's trucked by this rule.
What I need is implement other dial-pattern, I don't know if is better a free one that let dial any long dialing or more than 8 as rule.
So, what's the best rule or the best practice to implement into the dial-patterns that make my use and PBX setup easily ?
!!Outbound from SIP to Lines
dial-peer voice 21 pots
description FXO 0 Linea 1
preference 1
destination-pattern 9....
port 0/2/0
!
dial-peer voice 22 pots
description FXO 1 Line 2
preference 1
destination-pattern 8....
port 0/2/1
!
dial-peer voice 201 voip
service session
destination-pattern 201
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 202 voip
service session
destination-pattern 202
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
!! Inbound from Phone to SIP Server
dial-peer voice 2 voip
destination-pattern ........
session protocol sipv2
session target sip-server
dtmf-relay sip-notify
codec g711ulaw
no vad
Best Regards!
