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PRI/SIP Trunk for Outbound Dialing


I have someone who wants to do the following.

All Inbound from PRI

Outbound up to 10 calls Via SIP Trunks, outbound after 10 SIP Calls to PRI.

For this kind of config I'm going to have to use H.323 on my Gateway correct? No way to use MGCP, correct?

Has anyone done this before? I've read up a bit on the SIP part and it looks like I'd set them to Preference 1 with a max calls equal to the number of trunks purchased? and then the PRI as pref 2?

I've also been seeing things about CUBE and how it makes things better with SIP Trunks. Does this apply in my scenario?

Thanks in advance!


Re: PRI/SIP Trunk for Outbound Dialing

I think your call flow is this:

CUCM--SIP or H323--CUBE---PRI and SIP trunk

For the outgoing SIP trunk all you need is max-conn 10 on the outgoing SIP dial peer.

From there, you can let CUBE route the call directly to the PRI without re-routing it back to CUCM. Just put the pots dial peer to preference 1 or higher - 0 is the default.

If you're only using CUCM, this would be a combination of a route-group and something called "Call Admission Control". It's kind of a detailed topic, so I would suggest some googling. CUBE would make this easier, yes. Among the other reasons to use CUBE.


New Member

Re: PRI/SIP Trunk for Outbound Dialing

Hi, Nick,

We have the same situation, I plan to run SIP trunk between CUCM and CUBE, CUBE has both PRI to PSTN and SIP trunk to ITSP. Inbound PSTN calls are always routed to CUCM via MGCP. For outbound calls, can I simply route all outbound calls from CUCM to CUBE? or I should still configure SIP and MGCP gateway in CUCM's outbound call route-list with SIP trunk having higher precedence? the reason to do the latter is that if SIP trunk is not available (either the SIP trunk between CUCM and CUBE, or the SIP trunk between CUBE and ITSP, or both), we would still want to route outbound calls to PSTN, although the SIP trunk between CUCM and CUBE is highly unlikely to go down.



Re: PRI/SIP Trunk for Outbound Dialing

Hi Jian,

MGCP is a different case from H323. With SIP and H323, once the call is sent to the gateway it will make all the decisions. This isn't the case with MGCP, so all calls that go out of the MGCP PRI have to go through CUCM first. Meaning, if you would like to fail your SIP trunk back, it will need to go back to CUCM.

You would have both the MGCP gateway and SIP trunk defined in CUCM, and use Route Lists / Groups to define the redundancy.