Asterisk and Cisco Router FXO Incoming Call

Cisco routers can be used as a voice gateway for your Asterisk PBX. In this lesson I’ll show you how to configure your Cisco’s FXO port so that it will forward PSTN calls to Asterisk.

Cisco Router

The first thing to do is configure our Cisco router so that it will forward calls from the PSTN to Asterisk through SIP. Let’s start with the voice port:

voice-port 0/3/0
connection plar 500

Now we will create a dial-peer so that the calls are forwarded to Asterisk:

dial-peer voice 500 voip
 destination-pattern 500
 session protocol sipv2
 session target ipv4:<ip of asterisk>
 codec g711alaw
 no vad

This is all you have to do on your Cisco router. Let’s move on to Asterisk…

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Thank you for sharing. Is it possible to configure the Cisco UC500 to connect to a VOIP GSM gateway (e.g. Portech MV-374)? How to configure it on the Cisco side such that you have four (4) trunks created each presenting a SIM card? If you any idea I will grateful if you can share it.

Thanks

Hello Jorge,

You can use a Cisco access-list for this. By default Asterisk uses SIP on port 5060 and I believe UDP port 10000 - 20000 for RTP traffic. An access-list outbound to the Asterisk server could look something like this:

Router(config)#ip access-list extended ASTERISK
Router(config-ext-nacl)#permit udp 192.168.1.0 0.0.0.255 host 192.168.2.1 eq 5060
Router(config-ext-nacl)#permit udp 192.168.1.0 0.0.0.255 host 192.168.2.1 range 10000 20000

In this example, 192.168

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Hi Doug,

It’s been awhile since I configured this. I’m running Asterisk 1.8, here’s my config:

sip.conf

[my-trunk]
type=friend
host=x.x.x.x
context=fxo
insecure=very
qualify=2000

And extensions.conf:

[fxo]
; my-trunk
exten =&gt; 104,1,Answer
exten =&gt; 104,n,MusicOnHold(mp3)

This works for me.

Rene

Right now, my cisco uses plar 100 to get to the sip session and destination pattern 100 and session target ipv4:xxx.xxx.xxx.xxx to poing to asterisk (debian7 apt default version). My dial plan has exten => 100,1,Answer ()

This works as long as I use clid strip in my cisco dial peer voice voip. As soon as I remove it calls go nowhere.

This may be just a very poorly documented sip mangling by cisco that is causing this (not sending clid info in the correct setup message) or it is how asterisk deals with clid’s.

By the way, this site has helped me understand more

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Hello,
I’m having problems with my outgoing settings.

Here are my settings of a Cisco 2811 router:

voice-port 0/1/0
trunk-group 1 1
supervisory disconnect dualtone pre-connect
supervisory answer dualtone
input gain 10
output attenuation -1
no vad
no comfort-noise
cptone AR
connection plar 400
description (54) 11-4922-5216
caller-id enable
!
dial-peer voice 1 pots
description Linea 541149225216
preference 1
destination-pattern 9T
port 0/1/0
forward-digits 8
!
dial-peer voice 400 voip
numbering-type unknown
destination-pattern .T
session protocol sipv2
session ta

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