[Asterisk-Users] Asterisk/Cisco Call Manager 3.3

[Paul Davidson]

>
>
> Message: 13
> Date: Fri, 14 Oct 2005 09:58:37 -0500 (CDT)
> From: <gorand at dvvti.com>
> Subject: [Asterisk-Users] Asterisk/Cisco Call Manager 3.3
> To: <asterisk-users at lists.digium.com>
> Message-ID: <4408.69.215.189.2.1129301917.squirrel at www.dvvti.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> I need to pick all the Asterisk and Cisco People a little.
>
> My company has a Cisco Call Manager 3.3, configured via h323 gateways. I
> have remote users that I want to place a SIP Server on the external WAN
> and be able to connect their phones to the system and be able to get calls
> and call people in the office going through the Cisco Call Manager and the
> h323 router. My only problem is that Cisco Call Manager 3.3 does not
> support sip trunking. Is there anyway this can be done.
>
> Please shed some light on this topic.
>
> Thanks.
>
> Goran
>
> Goran-

Speaking from experience, you have a tough road ahead of you. The only way
to accomplish this is via h.323 trunks under Cisco and Asterisk. There are a
few known good configurations- I can really only speak to one, as it
eventually worked for me- but others may have different and perfectly
reasonable advice.

First, some prerequisites:
1. Asterisk 1.2 or CVS HEAD. Do NOT try this with any of the 1.0X series-
you will be able to call from CCM to Asterisk, but not from Asterisk to CCM.
2. An H323 Gatekeeper. GnuGK works, but does occasionally bonk out. CCM will
send RRQ requests to the gatekeeper at a rate of 10x per second, and
eventually, GnuGK loses it. An IOS gatekeeper seems to be much better.
3. chan_h323 set up and running properly. There's whole readme files on the
prerequisites for this- read them, follow the directions closely- and call
on JerJer *LAST* if you value your life.
4. A Gatekeeper controlled Trunk on CCM. The tricky bits here are the
significant digits, and the technology prefix. CCM does *NOT* register the
tech prefix or it's extensions with the gatekeeper- so you'll have to config
the gatekeeper to know where to send the call, and you'll have to configure
your CCM dialplan to act accordingly.

Set this up slowly. Get a working Asterisk box that's able to handle
softphones or hardphones as an island PBX, then configure the H323 trunk-
you'll save some frustration of trying to configure both simultaneously.

Find me on the IRC channel if you need specific questions answered- or email
me directly. I can optionally configure it for you for a fee- I'm based in
the US, and judging from your accent, I'd say you aren't- I can do this
remotely if needed. I won't charge you for questions answered. :)

-pbd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051014/49ea78c6/attachment.htm