I’ve been playing with Asterisk lately, and needed to get out to the PSTN. Since I had a Cisco AS5350 with a PRI attached for incoming RAS, I figured I could use that. I also had some DIDs on the line that I could use to separate voice and data calls.
Googling around, I couldn’t find a good example of anyone who has done it, so I was on my own.
My goals here were simple:
- People dialing 555-1111 should be picked up by the AS5350’s internal modems
- People dialing 555-222X should be forwarded to my Asterisk box for handling
- The Asterisk box should be able to dial out using the PRI
Already the DIDs were assigned to this PRI, so that wasn’t a problem.
The configuration on the AS5350 is fairly straightforward once you get it working:
voice service voip
! Bind the SIP and RTP sessions to a known interface
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
! Normal T1-PRI configuration using all channels
controller T1 3/0
cablelength short 133
pri-group timeslots 1-24
! Anything not picked up by a dial peer defaults to a modem call
isdn incoming-voice modem
! Use a data dial-peer to answer anything to 555-1111 with a modem
dial-peer data 10 pots
incoming called-number 5551111
! Match inbound calls to the voice DIDs
dial-peer voice 100 pots
incoming called-number 555222.
! All digits are collected and sent along
! The IP leg of the call... send to the sip server defined below
dial-peer voice 200 voip
session protocol sipv2
session target sip-server
! Any digits collected after the call has been answers get forwarded
dtmf-relay sip-notify rtp-nte
! I want g711 ulaw for a codec
! Anything else gets sent out the PSTN
! Note you need the data dial-peer above to accept modem calls, otherwise
! this dial-peer will give dialtone to anyone who calls!
dial-peer voice 1000 pots
! define your SIP server
authentication username USERNAME password PASSWORD
! xxx.xxx.xxx.xxx represents the Asterisk box
registrar ipv4:xxx.xxx.xxx.xxx expires 3600
After that, it’s just a matter of configuring a sip peer in /etc/asterisk/sip.conf