Routing DID between MGCP and SIP on the Same Router

[ibrahim.nezar]

Hello,

We have CUCM  connected to a gateway using MGCP:

CUCM -- VG -- E1 -- Provider

And a range of DID from the provider let's say (551101 - 551115). And this range (all of it) is passed to CUCM through MGCP upon incoming and outgoing calls.

We want to create a SIP trunk on the same gateway toward another CUCM and dedicate one number from the above DID range and bound it to this SIP trunk. So what we imagined is that we need to make the SIP config on the gateway and specify the dial plan and give it for example 551105. Now if any incoming calls hit this number, it should be routed through this SIP trunk to the other CUCM.

The question is, how the voice gateway decide whether to route this number toward MGCP or SIP? Do we need to separate or exempt this number from MGCP, so when a call toward this number hit the gateway, it will be be directed toward the SIP trunk instead of MGCP.

Another question, do you think the setup we are trying to do is good or there is a better way

Appreciate your advice

[Shalid Kurunnan Chalil]

Hi there, 

Edit: updated  the wrong option suggested (2)

I believe you are referring to two different call manager clusters here.  In this case, you have multiple options to achieve this. 

1. Create an Intercluster trunk between the call manager cluster 1 and call manager cluster 2 and route every call to the number 551105, to the call manager cluster 2. (minimal configuration)
2. IF you are choosing the SIP trunk from the gateway to Call manager cluster 2, then dial peer will decide where to send the calls. for example, (basic config reference)  - Need to convert from MGCP to SIP (trunk)or H323 (gateway) . In both cases you have to introduce dial peer configuration on your gateway to route the call to either of the call manager . 

 dial-peer voice 100 voip
description *** Towards Call Manager *******
preference 1
destination-pattern 551105

session target ipv4:X.X.X.X (callmanager2ip)

session protocol sipv2
session transport tcp
dtmf-relay sip-kpml sip-notify rtp-nte
no vad

[b.winter]

Hi @Shalid Kurunnan Chalil,

your second option won't work, because when using a MGCP-controlled ISDN interface, the call will always hit the CUCM. It's then like a "physical expansion" of CUCM.

So, the dial-peer will never be chosen.

The second option would probably look more like this, when using the VG as a gateway between the CUCM's:

Provider -- E1 -- VG -- CUCM 1 -- VG as SIP -- CUCM 2

[Shalid Kurunnan Chalil]

@b.winter @Spot on. 

Apologies for the wrong info.

first option is the right choice if there are no other sever concerns. 
otherwise , you have to change to use the gw as either sip or h323.

apologies again @ibrahim.nezar 

regards,

regards, 

[Roger Kallberg]

H.323 is being deprecated in IOS, so that would not be a good option. SIP would be the preferred choice.

[Roger Kallberg]

Like @b.winter wrote the dial-peer would not be at play with MGCP. All the calls that comes in on the PRI, or whatever circuit there is, that is controlled by the CM via MGCP will be sent to that CM.

@ibrahim.nezar What you ask about is not feasible with MGCP in the mix. What you can do is to use SIP as the control protocol for your voice gateway communication with CM instead of MGCP. It's not all that hard to go about, it's more or less a matter of creating the needed dial-peers on the gateway to handle the call routing and then create a SIP trunk in CM that points to your gateways.

[ibrahim.nezar]

Hi @Shalid Kurunnan Chalil 

Thanks for your replies.  Well we also thought of option 1 as well, something like this:

Cluster 2 -- SIP Trunk-- Cluster 1-- MGCP -- Gateway -- E1 -- PSTN

Let me brief you more about our setup:

We have UCCX application on cluster 2 (with trigger number 1001) and we want to reach it when an incoming hits numbers 551105 at the gateway. So the call will go from the gateway all the way to cluster 2 and dial the trigger 1001.

1. We will establish a SIP trunk between the two clusters.

2. In CUCM cluster 1, we will create route pattern for incoming toward 551105, this RP will route the call through the trunk based on the route group/list.

3. In CUCM cluster 2, we have a translation patter for 551105 that transform it to extension 1001 (i.e the trigger).

For the trunk type between the two clusters, we will use SIP Trunk based on the recommendation.

Appreciate your advice

Best

[b.winter]

Sounds good from my point of view.

[Shalid Kurunnan Chalil]

Hi Ibrahim, 

Good one. Test and let us know in case you hit any related issues. 

Regards,