Hi,
I’m plainning to install Asterisk in my office and would like to hook it to the telephony network by using a voice gateway which is already installed (A Cisco router connected to a PBX over a PRI connection).
Is that possible ?
What configuration is needed on the Asterisk PBX to route calls to the voice gateway ?
Thanks,
Roy
If the cisco router speaks sip it is probably a farely simple config. If it speaks skinny it is probably more complex.
Matt
So what is the simple configuration you are suggesting ?
Roy
If you look at the sample config file, sip.conf, you will see examples of sip gateways. If you need further help don’t hesitate to ask. Digium also offers express support in this area if you are in need of quick assistance.
this config works for me and a lot of my customers
dial-peer voice 101 pots
destination-pattern 0.
direct-inward-dial
port 0/0
!
dial-peer voice 200 voip
destination-pattern 334…
session protocol sipv2
session target ipv4:192.168.20.10
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 102 pots
destination-pattern 0.
direct-inward-dial
port 0/1
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:192.168.20.10
in extension.conf, you need to create a rule something like this
[pstn-outbound]
exten => _0x.,1,Dial(SIP/${EXTEN}@<ip adress of cisco)
every dial which start with 0 gets routed to gateway
and in sip.conf, you need to create the router something like:
[gateway]
type=friend
context=
host=192.168.20.250 (ip address of cisco)
dtmfmode=RFC2833
disallow=all
allow=ulaw
enjoy,
joash