CME and Asterisk part 3, end of the trilogy?

A couple of years ago I went through a couple of iterations of CME to Asterisk integration. Since then we’ve been running fairly reliably with the Cisco 1760 router behaving purely as a SIP based gateway and Asterisk managing everything else. I’ve run our Cisco phones as SCCP using the chan_sccp on Asterisk.

Due to some recent purchases and the fact that I don’t have ACPI figured out on the laptop Asterisk is running on, I’ve decided to change the configuration a bit. I need the memory and the charger from that laptop so that version of Asterisk has got to go.

What I’ll be doing is going back to the 1760 as CME and not just the voice gateway. There was always something a little funky about chan_sccp on Asterisk so having the call control back in the hands of Cisco should work better. In trying to get to the most current version of CME on the 1760 I’ve discovered something. I have 96/32 for the memory/flash and the “recommended” configuration is 128/64 for CME4.0 and later. As far as I can tell CME4.2 isn’t supported on the 1760 at all. I was tempted to try it but that IOS version wasn’t available and the T train doesn’t seem to have progressed enough to make it by with that alone. So, I tried CME4.1 since 12.4(15)T is available for the 1760. It works! So far! I’ve only attached a single phone to it and there was no way I could copy the entire CME installation over. If I had a diverse mix of phones I’d be out of flash space but I only needed the files for the 7940/60 and the 7920 so it fits. Barely.

I spent the better part of last night getting CME configured again on the 1760. Phone registers, no problem. I set up AsteriskNow virtually and connected an IAX softphone to it. Also, no problem. After some tweaking of the Asterisk config and reminding myself how context’s work I was able to make calls back and forth between a 7940 registered to CME and the IAX softphone. I then got the voicemail button working on the 7940. Also, not a problem.

Now I’m stuck. I can’t get the MWI to work. Voicemail works, no problem. Just doesn’t seem to be sending the MWI to the CME. Even running a debug on the CME shows no activity related to the MWI. So, I’ve got something misconfigured on Asterisk.

I’ll continue to poke at the MWI problem. Once I have a working config I’ll post the details so maybe someone else can benefit from this.

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8 thoughts on “CME and Asterisk part 3, end of the trilogy?

  1. Hi,

    I saw some intersting info on about MWI.
    You need to include the following in your Telephony-service configuration (for CME):

    mwi sip-server unsolicited

    The keyword here is UNSOLICITED.

    Anyway, I’m having some problems and I’m wondering if you could help??

    I’ve setup tow 7960 on a Cisco 3725 running CME 3.3. I’ve also setup Asterisk and some phone (analog on ATAs and X-lite on 2 laptops).

    I can make calls from Asterisk to the CME without any issue, however I can’t seem to get the CME side to ring Asterisk.

    Any ideas??


  2. Ayo, more often than not that’s a problem with applying the proper context on Asterisk. The outbound dialing is affected by the context the CME extensions are defined in, which would probably be related to a SIP trunk between the two. The Asterisk phones themselves may be in a separate context and the SIP trunk inbound would need to be included in that context or the call would fail. Context’s are kind of like a permissions system for what can dial where in the Asterisk world.

    Also, look up how you can debug the Asterisk. I don’t recall the exact command but you want to go in in verbose mode. The debug console will give you a lot of info and has always been key in me troubleshooting dial plan problems.

    Same goes on the Cisco router. “debug ip voip ccapi” (I think) provides a lot of detail of how the CME is interpreting it’s dial plan.

    Good luck!

    Oh, and BTW- the problem I’m at with Asterisk sending back the MWI isn’t with CME but with Asterisk. The way CME wants the MWI light lit isn’t the same as the SIP standard. Because it’s a unique way it requires some custom coding on the Asterisk and I decided I wasn’t quite ready to dive into that.

  3. Thanks for the info, I really appeciate it.

    I got it working shortly after making the post on your website. I can now make calls both ways.

    I changed the host statement in my SIP.conf file from host=dynamic to host=x.x.x.x and that fixed it.

    Thanks once again!!!


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