Paging setup with Cisco 4331 and NIM-2FXO Voice Analog Module

[BWeed2112]

First time posting, so bear with me here.

We currently have a Cisco 2911 with a VIC3-4FXS/DID card which is connected to a Valcom V-9940.

The Config on the router for Paging is:

dial-peer voice 200 pots
description *** PLANT1 PAGER 60 ***
destination-pattern 60$
port 0/1/0

A year or so ago we upgraded our router to a Cisco 4331 ISR that has a NIM-2FXO Voice Analog Module, and were informed our current Valcom V-9940 was not compatible with FXO, so we purchased a Valcom V-2000A to accommodate this.

At the moment we are still using the 2911 solely for paging, Call Manager is setup to send the paging ext to the IP of the 2911.

The other day we set Call Manager to go to the new ISR and ran new wires to V-2000A. When we tried calling the paging extension it gave us a "The number can not be completed" error.

I believe this to be an issue with the config on the 4331 as it is set exactly the same as the 2911.

Does the new FXO card need different commands to setup? For reference the 4331 Config is:

dial-peer voice 200 pots
description *** PLANT1 PAGER 60 ***
destination-pattern 60$
port 0/1/0

Any suggestions and guidance is very much appreciated. Hopefully we can figure this out

[Vaijanath Sonvane]

Hi,

I have a similar setup between Cisco 4351 FXO Ports and Valcom V-2001A. The configuration of Cisco 4351 depends on what protocol you use between CUCM and Cisco 4351. In my case, I am using Cisco 4351 as a CUBE Router as well as for paging integration, so I have SIP Trunk configured in CUCM. Below is the Cisco 4351 configuration for paging:

!
voice service voip
 no ip address trusted list
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 supplementary-service media-renegotiate
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  header-passing
  error-passthru
  options-ping 60
  no update-callerid
  early-offer forced
  privacy-policy passthru
!
voice class codec 1
 codec preference 1 g711ulaw
!
voice class server-group 1000
 ipv4 10.10.100.21 preference 1
 ipv4 10.10.100.22 preference 2
 description ** CUCM SERVER GROUP **
!
interface GigabitEthernet0/0/0
 description ** CONNECTED TO LAN **
 ip address 10.10.100.5 255.255.255.0
 negotiation auto
!
voice-port 0/1/1
 description ** CONNECTED TO OVERHEAD PAGING **
!
dial-peer voice 1011 voip
 description ** INBOUND PAGING CALLS FROM CUCM CLUSTER **
 session protocol sipv2
 session server-group 1000
 incoming called-number 10491$
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 10491 pots
 description # OUTBOUNG CALLS TO OVERHEAD PAGING  #
 destination-pattern 10491$
 no digit-strip
 port 0/1/1
!

In CUCM, I have following configuration:

1. Partition and Call Search Space for Paging Access
2. SIP Trunk pointing to Cisco 4351
3. Route Group with Paging SIP Trunk, Route List and Route Pattern

Also, make sure you have correct cable connections and switch positions on the Valcom device. Here are from my setup:

Only Brown and Blue Cable Used for ConnectionOnly Brown and Blue Cable Used for Connection

[BWeed2112]

Thanks for the quick reply. Based on the config you sent over i will send over what i have for those same sections.

I think this may have been me being dumb and plugging the rj11 cable into the wrong physical port on the 4331 (0/1/1 instead of 0/1/0). From what you see from the current config, if i plug things into the correct port, should this work? Once i get this working i will also add descriptions to the ports for easier visibility down the road.

Is it necessary to have the Battery Feed on the Valcom box set to On?

Let me know if you need any additional info to further assist, and if there is anything in CUCM that needs to be verified. (If that is the case if you know where i need to go in the GUI to find it that will be helpful as i am still learning my way around)

4331 Config

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323

!
voice-port 0/1/0
no battery-reversal
bearer-cap Speech
!

dial-peer voice 200 pots
description *** PLANT1 PAGER 60 ***
destination-pattern 60$
port 0/1/0

[Vaijanath Sonvane]

Hi,

Make sure you have below H.323 related configuration on your router:

voice class codec 1
 codec preference 1 g711ulaw
!
voice class h323 1
  h225 timeout tcp establish 2
  h225 timeout setup 2
!
interface GigabitEthernet0/0
 description # CONNECTED TO LAN **
 h323-gateway voip interface
 h323-gateway voip bind srcaddr <IP_Address>
!

If you have this configuration and if you connect RJ11 to correct port then you will require below additional configuration to make it work:

!
dial-peer voice 200 pots
 description *** PLANT1 PAGER 60 ***
 destination-pattern 60$
 port 0/1/0
 no digit-strip
!
dial-peer voice 1011 voip
 description ** INBOUND PAGING CALLS FROM CUCM CLUSTER **
 incoming called-number 60$
 voice-class codec 1 
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 no vad
!

 I had to turn on battery feed as paging was not working with it.

[BWeed2112]

Looking over the router config, i do not show any of the Voice Class lines, and the line that you show for GI0/0 is setup on a loopback interface

interface Loopback0
ip address 172.16.101.1 255.255.255.255
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.101.1

I want to make sure that adding the lines of the config that you sent will not break any of our existing phone settings.

Is there a reason why these lines of code are necessary when they were not required on the old router?

[Vaijanath Sonvane]

Hi,

When you configure Cisco Router with H.323 protocol then you need those commands to communicate with CUCM using H.232 protocol. Take a look at this video:

https://www.youtube.com/watch?time_continue=60&v=3BqLzlI4DxQ

[BWeed2112]

We were finally able to get back to this and with getting some additional outside help, this is the config we have that is working

voice-port 0/1/0
no battery-reversal
echo-cancel coverage 32
timing hookflash-out 50

!
dial-peer voice 200 pots
description *** PLANT1 PAGER 60 ***
destination-pattern 60$
port 0/1/0
!

It does not appear the "No digit-strip" or the " Dial-Peer voice 1011 voip" (and its config", are needed.

The only thing that doesn't work like the old system is when calling the paging extension, when it connects there is no audible "click" on the handset to notify the caller that they can begin speaking. The only thing that happens is the desk phone says "connected".

Is there any way for there to be an audible sound that will notify the caller when the call connects with this setup?