Cisco FXO as Trunk for Asterisk

Posted: January 24, 2011 by gngtech in Asterisk, Cisco, SIP
Tags: , , ,

Do you have a FXO card for your Cisco 1760, 2600, 3600?  Do you have Asterisk and want to use the FXO card as a trunk?  We answered yes to both of these questions and then spent hours trying to make it work.  Lucky for you we want to help, so here’s the quick and dirty on what we used.

Components:

  1. Cisco 2651XM with 128 DRAM and 48 Flash
  2. IOS 12.4 (IP Voice)
  3. NM-2V module in slot 1 of 2651XM
  4. VIC-2FXO card in slot 0 of NM-2V
  5. Line from phone company in slot 0 of VIC-2FXO
  6. Asterisk 1.6

Setting up the 2651XM

First off we need to get to the configuration mode so we can put in configuration commands.  If you’re familiar with Cisco you probably already skipped passed this wondering why we’d even bother describing these steps.  Simple answer: to be complete.

enable
conf t

So now that we’re in the configuration mode it’s time to configure the FXO interface.  More than likely you have a 2FXO card in your Cisco.  I’ve seen a 1FXO card option but I don’t quite understand why anyone would want one when they can have two.  Either way, we put our 2FXO in slot 0 of our NM-2V sled.  The 2FXO card has two RJ ports.  Through some simple experimenting we determined the right-most port to be a RJ14 (2-line port) and the left-most port is a RJ11 (1-line port).  You might think, “wow, does that mean I have 3 lines?”  Unfortunately, no.  The left-most port shares the second line from the right-most port.  Phones that support two lines through a single RJ14 can use the right-most port with a single cord to get two lines, while phones that support only one line can use either port without worry that connecting another one-line phone to the other port will lead to a shared line.

Cisco VIC-2FXO Manual Diagram

VIC-2FXO image from Cisco

  1. voice-port 1/0/0
  2. description FXO
  3. connection plar 603


  1. dial-peer voice 603 voip
  2. destination-pattern 603
  3. session protocol sipv2
  4. session target sip-server
  5. session transport udp
  6. dtmf-relay rtp-nte
  7. no vad
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