Using Cisco 2811 as Gateway between PBX and Asterisk/Trixbox

Unanswered Question
Feb 4th, 2009

I have an existing PBX that does not speak SIP. This PBX is connected to two PRI's that all are existing extensions are on. The PBX does speak MGCP and H323. Is it possible to use a Cisco 2811 with DSP's installed as a gateway to route calls from the PBX to the router and then from the router to my Asterisk/Trixbox server for my SIP phones ? Any suggestions on whether I should use H323 or MGCP between the PBX and router ? Does the router have the capability to receive H323 or MGCP, convert this to SIP , and send it to my Asterisk/Trixbox server ? Thanks.

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kelvin.blair Wed, 02/04/2009 - 14:23

yes a 2811 with the CUBE software can do everything you need. You can choose to use both hardware/software MTP's for this configuration as well. The configuration from CM to Cube should be H323 depending on your version of CM and out of CUBE to Asterick via SIP. This works with no issues.

bspear123 Wed, 02/04/2009 - 14:35

Can you explain what "CUBE" stands for ? Also, when you mention "CM" do you mean Call Manager ? I do not have call manager running, this is a third party PBX, not Cisco. Thanks !!

Nicholas Matthews Wed, 02/04/2009 - 15:00

CUBE, or IPIPGW will convert SIP-SIP, SIP-H323, or H323-H323.

It's a software upgrade but licensed product (paper).

MTPs are not applicable here because they are an artifact of CUCM which you do not have or use in this call flow.

Integration between two different systems is a common application. You can do many things on CUBE regarding translation, security, header passing/blocking, etc.

Doing a search for "Cisco Unified Border Element" will bring up much more information.



kelvin.blair Wed, 02/04/2009 - 14:39

I apologizeo on the type PBX. What type of PBX do you have? if it supports H323 then it will work the same way. The CUBE is Cisco Unified Border Element/IP2IP Gateway. Here is the link:')">

It is an IOS upgrade that will activate the CUBE, but you will have to pay for the upgrade. It is not covered by smartnet.

bspear123 Fri, 02/06/2009 - 14:21

No problem. The vendor of PBX is called Vertical Instant Office. It also supports MGCP. Would MGCP be a better choice than H323 ? If i implement H323 do I need to setup a Gatekeeper or will this work without a Gatekeeper ? Thanks !!

Nicholas Matthews Fri, 02/06/2009 - 14:53

A MGCP call agent is only supported on something like a BTS or on CUCM. A gateway / CUBE will not control another gateway. The gateway is the controlled side, not controlling.


bspear123 Fri, 02/06/2009 - 19:15

Ok. So I have the IP to IP gateway IOS software. I will be using H323 between the Vertical PBX and the Cisco 2811, and then SIP between the Cisco 2811 and Tribox Server.

I still need to know if I have to configure the Cisco 2811 as a gatekeeper, or just a gateway between the PBX and router??

Anyone have any example configs to assist me with my project ?

Thanks again !!

Nicholas Matthews Fri, 02/06/2009 - 19:21

This is the configuration guide for CUBE SIP-H323:

These are the essential global commands:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip



Then, whatever dial peers you need to configure.

'debug voip dialpeer' and 'debug voip ccapi inout' are helpful debugging commands.

Here is an example with CUCM SIP-H323:



Wes Schochet Sun, 02/08/2009 - 18:59

Why not go h.323 from the PBX directly to the asterisk server and leave the 2811 out of the whole thing?

Nicholas Matthews Sun, 02/08/2009 - 20:20

This is mostly because CUCM isn't nearly as flexible as CUBE for SIP. There are a lot of things like privacy, cause code mapping, NAT, header passing/modification, SIP REGISTER messages, and a number of other features that CUBE adds the CUCM just can't do (or at least as well).

This isn't so much the case for H323.


bspear123 Mon, 02/09/2009 - 06:37

I tried going H323 from my PBX to Trixbox, and could never get H323 to work on Trixbox. The Asterisk/Trixbox server does not support H323 on a standard install, only SIP. I could not get the H323 package to install error-free on my Asterisk server, so that is why I thought it would be easier to have the Cisco 2811 do the H323 for me.

bspear123 Tue, 02/10/2009 - 17:12

Can anyone provide some example dial-peer configurations? I am assuming I will need 2 dial-peers configured on my router, one to match calls from the PBX on H323 to send using SIP to Trixbox, and the other dial-peer to send calls from Trixbox using SIP to the PBX using H323 ?? Any help would be appreciated. Thanks.

Nicholas Matthews Tue, 02/10/2009 - 17:51

dial-peer voice 1 voip

destination-pattern 2...

session target ipv4:

session protocol sip

description SIP dial peer for 2000-2999

dtmf-relay rtp-nte

codec g711ulaw

no vad

dial-peer voice 2 voip

destination-pattern 1...

session target ipv4:

description H323 dial peer for 1000-1999

dtmf-relay h245-alpha

codec g711ulaw

no vad

Glad I could help :)


bspear123 Thu, 02/19/2009 - 17:36

I am still trying to configure this correctly. Maybe you could detail the dial-peers more for me ? I am attaching a diagram if that helps. I have SIP phones on Trixbox starting with extension 4000, the router is configured as a SIP peer with Trixbox. The legacy PBX has extensions that start in 200. How do I configure the router to receive calls from the SIP extensions on the Trixbox (4000) and then forward the call out to the legacy PBX via H323?

Nicholas Matthews Thu, 02/19/2009 - 20:17


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