Hi,

I hope it helps;

http://www.cisco.com/en/US/docs/routers/access/interfaces/ic/hardware/installation/guide/voi_wic.html

you are right, the problem is due to the IOS on your router, for Cisco2600 the default Flash is 8MB and default DRAM is 32MB and this is for all Cisco2600 series, now the problem is that the image on your router (c2600-js-mz.122-16b) need the Flash to be as minumum 16MB and the DRAM to be 48MB to supported vic2-2fxo card to work. am sure from this and this link support my idea, please check it
http://www.cisco.com/warp/public/cc/pd/rt/3600/prodlit/c36p_ds.pdf

BR

Hi,

Thank you for your replies!

The flash is 16MB on my router, and is working as per normal. The one thing that I do not seem to understand is why is my 2800 detecting the card and the 2600 unable to detect? Then I have the question, with my 2800 detecting the card is unable to show any voice ports?

ok, please use the following command on cisco 2800, you need it to enable the vic2-2fxo card configuration:
card type {t1 | e1} slot subslot
please confirm.

BR

will try this and get back to you. I have been told by some one that i need to enable telephony service which would then enable the voice ports?? I am not sure about this either, will have to check, tomorrow being the earliest.

Thanks

Hi BR,

Te command would work. It says unrecognised command. i can enable telephony service but I guess that is for Cisco Call Manager not for my Asterisk.

Still looking for a solution, will post back if I have a break through

Hi,

If I need to upgrade the flash from 16MB to 32MB do I need a SIMM or a DIMM flash? Might be a silly question but I am a novice. Also, my router has 61440k/4096k memory, do I need to upgrade memory as well if I upgrade the flash.

My router is a Cisco 2620XM with MPC860P processor.

Thanks

I still cant get this thing to work! Help!

This is what I am trying to do:
I have Asterisk running on one of my machines which is connect to a Cisco 2621XM router with an FXO card. The Router is connected to my internet gateway and the PSTN line comes through onto one of the voice ports of the FXO card.
I use Cisco 7960 phones to register with my Asterisk and call landlines within Australia or mobiles. I have the configuration given below.

Now, this worked for me, but I am not able to make out the change to the routers, I am not able to configure voice-port to get the configuration going.

Please help, someone...

CONFIGURATION:

26x0 Configuration:
First the FXO port config, set gain/attenuation/comfort noise as you wish, or omit entirely. The PLAR line determines what extensions inbound calls are routed too.
Note: the vic-2fxo, which is what I have, does not support CallerID, you apparently need the vic-2fxo-m1 for that.

voice-port 1/1/0
input gain 10
output attenuation 10
no comfort-noise
connection plar 8100


Now for the dial-peers
701 defines what to do when the router receives a request to dial a number that begins with 9, followed by any 7 digits, you could change this, but 701 will stop any LD call from being made... I have this that way so anyone who cuts and pastes will not arbitrarily allow long distance :) The forward digits helps enforce this.
dial-peer voice 701 pots
destination-pattern 9.......
port 1/1/0
forward-digits 7

Dial-peer 400 tells the router how to handle the PLAR extension that is defined on the voice port above

dial-peer voice 400 voip
destination-pattern 8100
session protocol sipv2
session target sip-server
codec g711ulaw

The Sip-UA defines where to send the calls for the sip-server, which is Asterisk, so set the IP to match your asterisk servers IP.

sip-ua
sip-server ipv4:209.X.X.Y


sip.conf
[general]
port=5060
bindaddr=0.0.0.0
;videosupport=yes  ; Gives an error: "process_sdp: Error in codec string 'ideo 0'"
disallow=all
allow=ulaw
context=bogon-calls

[y.y.y.y]
context=pstn-incoming
type=friend
host=y.y.y.y ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw

[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001

extensions.conf
[bogon-calls]
exten => _.,1,Congestion

[pstn-incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => pstn-outbound

[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to the PSTN
;exten => _9.,1,Dial(SIP/${EXTEN:1}@y.y.y.y) ; IP address of Cisco gateway
; 9 stripped by Cisco gateway
exten => _9XXXX,1,Dial,SIP/${EXTEN}@y.y.y.y ; IP address of Cisco gateway
exten => _9XXXX,2,Congestion

[lan-phones]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup

Can you submit a 'show diag'?  I'm guessing you don't have any DSP modules installed.

I think you need NM-1V Voice Interface Card to be used with the FXO card!

thanks