Asterisk integration with Cisco Call Manager

Asterisk Cisco Call Manager (CCM) Integration

Why integrate Cisco CallManager and Asterisk?

  • Features: Asterisk provides features that CallManager by itself does not.
  • Migration: Allow a gradual migration from a closed source PBX to open source PBX.

There are two ways to accomplish this:

  1. Using H.323: In CCM Asterisk appears as a H.323 Gateway.
  2. Using SIP (only in CCM 4.X+)

Here is instruction on how to integrate asterisk with Cisco Call Manager using SIP trunk.

Step1) On your freePBX create a new SIP trunk and set the following (remain the rest as is):

host=10.200.214.10
port=5063
fromdomain=10.200.214.10
fromuser=
username=
secret=
type=peer
qualify=yes
canreinvite=no
nat=no
insecure=port,invite
context=from-internal
disallow=all
allow=ulaw


Step2) you should define your outgoing calls to cisco

Step3)

  1. Open up the CallManager Administration web page.
  2. Since a SIP trunk requires MTP, make sure you have one:
    1. Service -> Media Resource -> Media Termination Point
    2. Normally your CallManager server should appear there if you do an empty query
    3. if not, go to the CallManager Serviceabilty web page, and activate the Cisco IP Voice Media Streaming App service
  3. Select Device->Trunk from the menu.

Image

Select the “Add a New Trunk” link from the upper right hand corner of the “Find and List Trunks” page.

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Select “SIP Trunk” as the “Trunk type” and “SIP” as the “Device Protocol”. Click on the “Next” butto

Image

  1. Enter a name in the “Device Name”. Valid characters are letters, numbers, dashes, dots (periods), and underscores. The device name is only used internally in Call Manager so it can be anything you want.
  2. Enter a description in the “Description” field.
  3. Select a device pool.
  4. Enter the IP address of your Asterisk server in the “Destination Address” field.
  5. Select “UDP” as the “Outgoing Transport Type”.
  6. Modify any other settings as needed for your (CiscoCallManager|CallManager) installation.
  7. Click on the “Insert” button.
  8. Add route patterns in CallManager that send calls to Asterisk using the SIP trunk that you just created.

Cisco Call Manager 6.1

It’ very simitar to 4.1, but you must change the UDP protocol of sip in this menu:

System > Security Profile > SIP Trunk Security Profile

Outgoing Transport Type: UDP

CM6SIPTrunkSecurityProfile2.jpg


Source) http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration

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