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CCM 3.3.6 and Asterisk


Has anybody integrated CCM 3.3.6 with the Asterisk open source pbx? I would like to use Asterisk as our voicemail server with CCM as our PBX, but I'm not sure to to integrate them. All of the examples I have seen use SIP and CCM 3.3.6 does not support SIP. From what I can find in CCM, I will have to use Asterisk as an H.323 gateway, but I have no idea where to start. Thanks

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Chris Deren
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You are correct Asterix is SIP system, where CCM 3.3 does not support SIP. So unless you create a h323 to SIP Gateway integration to link these systems it cannot be done. CCM 4.0 and up support SIP trunking, if you can upgrade to that release that would be your best bet.


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That's not true about *. It supports SIP, H.323, SCCP, IAX, etc. The problem with upgrading is that the upgrade license is $4000 per box. If I had a spare $8000 to upgrade two of them, I wouldn't need to use * for voicemail.

Well if it supports H.323, try creating h323 gateway on CallManager and point to *.

That's the issue. How do you "point to *"? There are basic config settings but nothing related to a gateway IP (at least that I can see). I'm assuming that CCM wants the gateway to register with it before it will send calls, but I don't know how to configure CCM to accept an H.323 registration from *.

You need to create GW. In Callmanager go to Device->Gateway->Add new Gatway->H.323 GW( second from last)

Enter IP of * as the device name, select all appopriate settings and Insert.

Here is Admin Guide for CCM 3.3(4)


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Not sure if you have figured out your solution yet, but here is another link off of This gives you a pretty decent rundown of how to configure both sides. I am not sure if you have any experience with Asterisk but this site is pretty much all Asterisk in nature. Some good tutorials and such

Hope this helps out. Depending on your experience with it, you may want to go with one of the live CD distro's that put on AMP (Asterisk Management Portal)

This is great information about integration between Asterisk and Callmanager for a SIP trunk using CM 4.X and above.

We are currently running Unity and maybe wish to migrate across from unity to *.

Is there anyone thats got this CM and * setup working? and has anyone came across any major or known caveat issues, while setting this up?

If lets say the CM or WAN link goes down, ip phone fallback to SRST mode. Will it just work exactly the same way with Unity was involved and route and retrieve calls via the PSTN?

Any useful information will be great!



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