Mapping Calls to FXO Ports on Analog Gateways

This can be found mostly nowadays with SRST configuration. But it is not only limited to SRST. If you have an ISR at home, you can do this with your home PSTN connection. I have a Cisco 2911 with FXO ports and some IP Phones which are registered to my VGW (Voice Gateway Router).

If you want to know how to register an IP Phone at a Router please click here.
If you want to know how to configure SIP SRST please click here.

Setup is simple, the RJ11 from PSTN analog line connects directly to the FXO port of the router. FXO (Foriegn Exchange Office) is a type of WAN interface card which an analog line from a PBX or a Telephone service provider can be connected to.




(1) Create a POTS dial peer for outgoing calls

VGW(config)#dial-peer voice 100 pots
VGW(config-dial-peer)#destination-pattern ..........
VGW(config-dial-peer)#port 0/0/0

Destination pattern .......... matches any number with 10 digits. port 0/0/0 is the physical analog port connected to the analog line.

(2) Configure an extension to ring for incoming calls

VGW(config)#voice-port 0/0/0
VGW(config-voiceport)#connection plar opx 1111
VGW(config-voiceport)#caller-id enable

This is configured to ring 1111 extension. Last command is to identify the caller's number.

Above configuration is enough to make calls out and receive calls in. Following command will show an active incoming call on port 0/0/0 as off-hook.

VGW# show voice port summary

Now let's say if you have 2 or more analog lines connected to several ports. You can create a trunk group and assign it to voice ports.

VGW(config)#trunk group PSTN
VGW(config-trunk-group)#hunt-scheme round-robin

VGW(config)#voice-port 0/0/0
VGW(config-voiceport)#trunk-group PSTN 1

VGW(config)#voice-port 0/0/1
VGW(config-voiceport)#trunk-group PSTN 2

In above configuration, Line choosing method is round robin which means if one line is busy it will go to the other line. 1 in the trunk-group sub command states it is the preferred line.

Note:-

show call active voice and debug voip dialpeer commands will be useful in troubleshooting.