Intercluster Trunk with 3rd Party PABX

[Siva chinap]

Dear All,
We intend to set up a SIP trunk between CUCM and a third-party PABX (Asterisk). The requirement is that Asterisk users make an external call through CUCM—> VoiceGateway—>PSTN.

Prior to deployment, I have some concerns.

1) Is it possible to get the exact did number if we configure the extension in the DID range in a third-party PABX when the call is routed through CUCM?

2) We intend to configure NAT on both CUCM and a third-party PABX. Will the SIP trunk be updated in source and destination with NAT IP, or do we need to provide the exact PABX destination IP address in SIP trunk?

[Roger Kallberg]

1) Yes you can pass the DID number.
2) I would recommend you to stay away from using NAT as it will bring you nothing but trouble. If you have a need to demarcate the CM from the traditional PBX I would recommend you to put a SBC (Session Border Controller), like a Cisco router running Cube services, in-between the two as then you’ll have the CM network connectivity on one interface and the PBX on another interface on the router. With this you’ll don’t need to mess around with NAT as the SBC would act as the border between your two systems and hide the network side of the two sides from each other.

[Siva chinap]

@Roger KallbergThanks for your swift response, In our scenario there is no SBC. Integeration is directly between CUCM and 3rd Paty PABX. WIll there be any issue if we use NAT IP on both side, is there any chance of call failures.

[Roger Kallberg]

Yes there is a big chance of that you’ll get issues with calls. NAT and SIP are not best of friends as you’ll need to modify the content in the various SIP headers in the SDP apart from the IP headers that NAT changes. With the setup that you describe there are no option to do that modification, that is if you don’t have a firewall, or something else, in-between the two that handles the NAT and have the capability to also alter the content in the SDP.