SIP Trunk Outbound Failover to ITSP for CUCM w/ two CUBEs
Having a lot of difficulty figuring out how to get outbound failover working in the event one of the CUBEs fail.
I have two SIP trunks from Verizon in separate data centers with one CUBE at each location. Each CUBE can also reach the other SIP trunk destination. (e.g. CUBE 1 Primary sip trunk destination is the sip trunk at that datacenter, but it can also reach sip trunk destination of sip trunk at data center 2)
They handle the PSTN failover inbound. I tested this successfully by creating a dial peer that matched a test number and set a translation-profile to block it, and the PSTN call came in through the other CUBE.
My issue now is trying to figure out how to get things working outbound to the PSTN. We have about 10,000 numbers that are split between the two trunks, so half of our DIDs are provisioned for SIP trunk 1 and the other half for SIP trunk 2. I have one route pattern configured for PSTN 9.[2-9]XX[2-9]XXXXXX pointing to a Route List that consists of CUBE-A, CUBE-B. What is causing problems is that Verizon doesn't allow the call to go out to the PSTN if the calling number is not apart of that sip trunk's did range. This means in the event that CUBE A goes down, if the calling number is associated with SIP trunk 1, it gets denied when the call is reverted to CUBE B, since it's not apart of CUBE B's trunk.
I tried two things:
1) created a secondary dial-peer in each CUBE that pointed to the sip trunk destination of the alternate trunk and attached preference 1. This worked but it took almost 20 seconds in a failover scenario for the call to go out. This is even after I configured sip retry INVITE timer to 2 and retry timer to 150.
so on CUBE A dial-peer 1 destination is 172.31.150.15:5075, and dial-peer destination is 172.31.150.15:5074
2) Tried to configure prefix on route list level and strip on the dial peers, but that didn't work.
Would greatly appreciate any insight. Diagram is attached
If I was in your shoes I would first engage Verizon Voip support, what's the point of having 2 Sip trunks if you won't have outbound redundancy? My ATT Sip trunks have no such nonsense...
Anyways, if the caller id is not crucial to your business, you could try to use an external mask when routing out the 2nd CUBE, you can add the mask at the Route List level. You may have to create a 2nd route group for this, I'll check my current config when I get to the office tomorrow.
Note: The external mask will be part of your Site A DID list but will only be activated when using the route group that points to CUBE B. Obviously your caller ID will reflect your external mask for all calls from Site A out of CUBE B.
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