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CUCM route calls diferents gateways/sip trunks

com
Level 1
Level 1

Hi at all, I have CUCM 6.1.1 and I want to route calls throughs diferents gateways or sip trunks.

I planned to do with route groups, but I can not add on a route group a H323 gateway and a SIP trunk at the same time.

How can route calls in different ways?

In the CUCM page "Route patterns" I want to make alternative routes, for example, the number 6666 is on route "666X" through a "gateway/route list", but if I can not contact by going this route I need to go through the alternative route "XXXX" through another "gateway/route list".

How can I make by going first to one pattern and then the other pattern?

Thanks!

Fran

7 Replies 7

Dennis Mink
VIP Alumni
VIP Alumni

You cant do this from call manager,

If your SIP trunk and H323 gateway are separate devices, you will need to stick a Gatekeeper in between. That GK can then make the decision to send the pattern out via SIP or via your H323 GW. From a CUCM point of view, you will need to build a trunk to your GK and send that whole 666x pattern across it.

Alternatively, terminate your SIP trunk on your H323 Gateway and manipulate the path taken, by means of dial peer prefernce

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Ok thanks but one question more.... if I have a MGCP Gateway? Can I do this from my MGCP Gateway? or I need an H323 Gateway.

And another possibility.... I dont know if it's right....

Can I do this with Partitions and CSS?

This is for example I 'll have a CSS "Global" with Partitions (Primary and Secondary);

It could go the route first to 666x Gateway with CSS "Global" and partitions (Primary and Secondary). This way I do not know if it is routed first through the Gateway of the partition as Primary and Secondary alternative partition that is served by the SIP Trunk.

Using the "Dial Number Analyzer" I get the second path XXXX (SIP Trunk) as an alternative route ...

You could do that. If your CSS contains two identical matches it picks the first partition that has a match (so make sure the order of the partitions is correct).

But please not that if you would use tha H323 as your primary, and your PSTN link goes down on it, the H323 would still be registerd and from a CUCM point of view still be a gateway that it can route calls to, rather then failing over to SIP.

With MGCP instead of H323 this is different,as CUCM is aware of the status of your PSTN line (E1/BRI), when this goes down, yes it will fail over to SIP.

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Ok but I think that it doesn't work because I want that when I make a call through the MGCP gateway and the remote destination are unknow or not available it automatically routing the call to another "route pattern" through Trunk SIP. The CUCM cannot know if the remote destination are unavailable. Is this ok? If it is correct I think that the only way to make this call routing through dial-peer on the MGCP gateway.

What do you think?

iventurini
Level 1
Level 1

You need to configure two distincts route groups (one for H323 gateway and one for SIP trunk) and then put them into a route list. You will be able to set an order to the route groups.

Hope that helps

Regards

Ignacio

Thanks for your answer!

When I try to put this route groups into a route list I get this error:

"Update failed. [1803] Se ha detectado una configuración QSIG no válida para esta lista de rutas

Route List Member Information

Route Group

Calling Party Transformations

Use Calling Party\'s External Phone Number MaskRequired Field"

Any ideas?

Regards,

Francisco.

One clue....the two route groups that I try to add to route list are "GATEWAYS[QSIG]2 and "TRUNK SIP [NON-QSIG]". If I put into the route list "TRUNK SIP [NON-QSIG]" first and then I try to add "GATEWAYS [QSIG]" I get the error UPDATE FAILED [1803], but If I put into the route list "GATEWAYS [QSIG]" first then when I go to add route group page I don't see "TRUNK SIP [NON-QSIG]" I only can see "Standard Local Route Group" that I don't know what it means.

Is there any problem with that configuration reported by Cisco?

Is it a system feature?

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