Legacy PBX integration - H323 - Dial-Peer separation

[Douglas Fernando Fischer]

One of our costumer is implementing IP Telephony.

The project includes an integration with a legacy PBX(TDM).
The idea is to used this PBX as a big VG.
Only calls between extensions inside the PBX will be manage by it, every thing else will be sent to the E1.

That E1 will be connected to a 2911, and because of the legacy it will use R2-Digital.

This 2911 will also be connected to the PSTN using an E1, and for our nightmare it will also be with R2-Digital.

The problem?
A- R2-Digital it not compatible with MGCP.

So, in the most basic  configuration possible, if some user in the PBX does a call to the PSTN,  that call will be routed directly to the E1 connected to the PSTN  without passing through the call processing of callmanager.


We have two costumers with this king of integration with legacy PBX,
But on those clients, the E1s run on ISDN-PRI and are configured with MGCP on call-managers.
Consequently all the calls flow through call processing of callmanager.


Why does the calls need to flow through calmanager?
A- CDR and Forced Authorization Codes.



Possible solutions
------------------

I was thinking about something using Class Of Restriction, but I didn't saw the full solution.

My Idea was create two LoopBacks and associate the calls those two E1s each one with each LoopBack.
On the CallManager, I would create those two loop backs as different gateways.


There is any suggestion?

[Marwan ALshawi]

Well if you make the gateway work as H323 with PBX and CUCM you can do the bellow work around at least H323 with PBX and might be defined as MGCP with CUCM

Let's say the numbers range of the PBX extensions is 5xxx

In the gateway create pots dial peer if the call coming via the PBX to the gateway using tdm not VoIP, if it's come as ip just use VoIP dial peer but the idea is to use an incoming dial peer with the following work around

This dial peer will match any calling number with range of PBX extensions example here 5xxx and prefix a digit to the called number here this digit will be 8

As mentioned above you might define the gateway using different loopbacks as h323 and MGCP in CUCM

Now create dial peer that match any called number start with 8 and send it to CUCM and call manager when the call comes via the gateway as H323 ( you can give it different CSS ) strip the 8 and leave the original called number to go through CUCM dial plan may be back to the pstn but now through CUCM and then the voice gateway

Example config of the dial peers

Voice translation-rule 1

Rule 1 // /8/

Voice translation- profile PBXin

Translate called 1

This  dial peer matches the calling number of PBX extensions using the command answer-address and add 8to the called number

Dial peer voice 8 pots

Answer address 5...

Translation-profile incoming PBXin

Now this is a VoIP dial peer that send all the called number start with 8 to CUCM

Dial peer voice 88 VoIP

Destination pattern 8.*

session target ipv4:CUCMip

Hope this help

If helpful rate

[Marwan ALshawi]

Did you try it ?

[Douglas Fernando Fischer]

Sorry, no.

This environment is not active yet.

We will prepare a lab environment to it.

In addition to your suggestion, I thought to ask to PBX guys to already send-me that extra digit at the beginning of calls destinate to PSTN. In this case I wouldn't need that translation.

In my opinion, the problem in this solution will become in SRST mode, where I will need to treat that extra digit also.

I need to think a bit about it.