H323 gateway with SIP trunk with CUCM

[jaheshkhan]

I want to implement voice gateway for one of the branch site . They have PRI connection. 

Can I use SIP trunk between CUCM and Voice gateway. then In VG, I configure PRI configuration and required dial peer for calls?

can I do that way?

What all configuration I have to follow to do above setup if that possible?

Since this is branch site, Im planning to implement SRST as well.

CUCM <-----> SIPTRUNK<-----> VG <----- >PRI LINE

[Mohammed al Baqari]

Hi,

My suggestion to hire a professional services engineer who has
experience in similar deployments to implement this correctly.

Here are some documentations.

https://www.ciscopress.com/articles/article.asp?p=2492950&seqNum=4
https://netcraftsmen.com/configuring-srst-on-cme/
https://community.cisco.com/t5/collaboration-voice-and-video/implementing-sip-gateways-in-cucm/ba-p/3100278

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[Roger Kallberg]

No need to mix in H323 in the setup. You can achieve all that you ask about with only SIP. There are plenty of documentation on cisco.com on how to setup a voice gateway with a SIP trunk to CUCM. For SRST it works equally well with either protocol.

Here is one starting point for gateway configuration. https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/config_library/15-mt/vg-15-mt-library.html

[jaheshkhan]

thanks. My question is that can I use one side SIP trunk ie CUCM to VG 

on the other side from VG to telcom provider just the PRI connection. I cant request client for SIP trunk at this moment. because they  have existing PRI line. 

CUCM to VG --> SIP TRUNK

VG to Telcom --> PRI configuration with dial peer.

This is part of migration of Nortel to CUCM ..

[Roger Kallberg · Accepted solution]

Yes that’s absolutely possible. It’s a standard setup with ISDN on the PSTN side, aka PRI for the connection to PSTN and then SIP on the VoIP side, aka to CUCM. This would be a classic TDM type of voice gateway.

[Nithin Eluvathingal · Accepted solution]

On CUCM,  voice gateway with PRI can be configured using   below

• SIP-- Dial Peers required on Gateway
• H323-- Dial Peers required on Gateway
• MGCP--- NO dial peer required