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You are here: Home / IP Telephony / Connecting two Cisco Unified Communication Manager Express with H323

Connecting two Cisco Unified Communication Manager Express with H323

Written By Harris Andrea

The Cisco Unified Communications Manager Express (CUCME) is the new brand name given by Cisco to the older Call Manager Express (CME) system. The concept is the same however: IP Telephony software running on Cisco routers. Therefore, the CUCME is a normal Cisco router (models supported are 1800, 2800, 2900, 3800, 3900 series) with a special IP Telephony software (call manager software) installed on the router’s flash memory. The CUCME system serves as the call control node to facilitate IP Telephony communications in a small to medium size Enterprise.

Usually there is a single CUCME system in each LAN network, with several IP phones connected on the LAN switches. An enterprise with several sites connected over a private IP WAN network can establish full IP voice communications between sites by configuring H323 communication between each CUCME router. A simple example with a two-node topology is shown below.

CME-A node has local IP phones with numbering 500x and a WAN IP address of 1.1.1.1. On the other site, CME-B has local IP phones with numbering 600x and a WAN IP address of 2.2.2.2. By establishing H323 voip communication over the WAN (between 1.1.1.1 and 2.2.2.2) we can have full IP telephony conversations between the IP phones of both sites.

MORE READING:  Comparison of H323 vs SIP Protocols Used in VoIP and IP Telephony

CAUTION: Because the actual VoIP RTP traffic communication between site A and site B will be running from one IP phone to another IP phone, there must be full IP routing established between the IP phone subnets.
The CUCME configuration to establish H323 between the two sites is shown below:

CME-A

CME-A#show running-config
dial-peer voice 6000 voip
destination-pattern 60..
session target ipv4:2.2.2.2
dtmf-relay h245-alphanumeric
codec g729r8

CME-B

CME-B#show running-config
dial-peer voice 5000 voip
destination-pattern 50..
session target ipv4:1.1.1.1
dtmf-relay h245-alphanumeric
codec g729r8

The dial-peer configuration on CME-A tells the system that in order to reach the destination pattern 60xx the session will be established with IP address 2.2.2.2 (i.e CME-B). The inverse applies for CME-B.

Note: Make sure to select one of the high compression codecs ( such as g729, g723) in order to save bandwidth for voice calls over the WAN network. Each VoIP conversation using a high compression codec (g729, g723) will use significantly less bandwidth compared with the traditional G711 codec.

Related Posts

  • Comparison of H323 vs SIP Protocols Used in VoIP and IP Telephony
  • How to Use Cisco ECC Profile to Provide Caller ID Details for External Calls
  • IP Telephony and VoIP Tutorial
  • Cisco UC560 Dial Plan for Voice Mail Configuration Example
  • How does VoIP work-Brief Overview

Filed Under: IP Telephony

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About Harris Andrea

Harris Andrea is an Engineer with more than two decades of professional experience in the fields of TCP/IP Networks, Information Security and I.T. Over the years he has acquired several professional certifications such as CCNA, CCNP, CEH, ECSA etc.

He is a self-published author of two books ("Cisco ASA Firewall Fundamentals" and "Cisco VPN Configuration Guide") which are available at Amazon and on this website as well.

Comments

  1. Raja says

    September 21, 2010 at 9:01 am

    When i trying to connect on GNS3 these conf are working but when i am trying on real network its not working….
    Plz help

  2. saeed says

    May 5, 2011 at 9:53 am

    thank you
    You save 2 work days for me! :)

  3. Daniel says

    February 21, 2012 at 4:04 pm

    When I tried this the codec didnt work I had to change it to g711ulaw. How come it didn’t work the other way? What is the difference in the codecs?

  4. Blog Admin says

    February 21, 2012 at 6:56 pm

    Daniel,

    I didn’t encounter similar problem before. The only I can think of is maybe the IOS or CME software that you are using does nor support high compression codecs such as g729

  5. Devin says

    May 19, 2013 at 3:56 pm

    What if I have CME to CME communication through a VPN tunnel do you still need to specify dtmf-relay h245-alphanumeric.

  6. Blog Admin says

    May 19, 2013 at 5:45 pm

    Yes Devin, you still need the dtmf-relay command. It doesn’t matter if you are connecting via VPN or other means.

  7. Rhonda says

    April 23, 2014 at 5:53 am

    I have just about the exact same layout, but with three more routers. I have been told that a dial-peer voice 10 voip
    incoming called-number .
    Must be configured also. Before I saw your post I was not getting anywhere, now I can call between phones on same router, but not between sites yet. I can ping everywhere, phones, routers, ip route finds the cme ip….
    Do you have any suggestions?
    Thanks.

  8. BlogAdmin says

    April 24, 2014 at 9:52 am

    Hi Rhonda,

    When you say three more routers, do you mean all of them having ip telephony software and participating in the IP telephony network? You will need to configure additional dial-peers to reach the other CME sites. Please give me more information about your topology and dial-plan.

    Harris

  9. ryan says

    May 17, 2017 at 7:50 pm

    Any one still viewing this please help.

    will this solution work over a firewall VPN to Firewall VPN and if so are you using all public IP addresses for each phone?
    can some one please help explain. I would like to use the two call managers separated by a VPN and use one T1 line.

  10. Harris Andrea says

    May 18, 2017 at 7:56 am

    Ryan,

    If you configure site-to-site IPSEC VPN between the two LANs, they will be reachable with their private IP address ranges, therefore the solution will work.

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