CUCM - Asterisk Integration

[gt364555]

I am trying to integrate our Asterisk PBX setup with CUCM where,

1. CUCM will be connected with the PSTN network

2. Asterisk will be handling IVR and voicemail with inbound SIP trunk from CUCM

3. For transfer of the call to an agent, the outbound SIP trunk from Asterisk to CUCM will be used to connect to the IP phone connected to CUCM.

I have used the below tutorial as a reference for the above integration setup,

https://www.uccollabing.com/cucm-asterisk-trunk-integration/

I have one question though as I am trying this setup for the first time,

  Does the transfer of calls from Asterisk to CUCM using the above setup work as expected? Or are there any possible problems with this setup?

It would be really helpful if anybody can guide me with this setup.

Thank You.

[Ritesh Desai]

Hey Hi,

If you will be using CUCM only for agent transfer after the IVR treatment then in that case you need to make sure your SIP TRUNK IS UP on CUCM and between Asterisk. SIP is UP for xx days xx minutes this is visible on device > Trunks >

in order for SIP trunk to be UP for successful routing of calls, you need to make sure SIP PARAMETERS are same on both the ends.

eg: sip transport tcp or udp, sip port no, dtmf signalling, security, etc. You can set this in SIP Profile under Devices and SIP security profile under System.

[gt364555]

Thanks Ritesh for quick response and also pointing out about the SIP parameters!

[Nithin Eluvathingal]

I have done exactly the reverse. 

• Asterisk gateway used for connecting PSTN
• Asterisk integrated with CUCM using SIP Trunk
• New users where using CUCM
• old users, and their voicemail   where in Asterisk 
• call center agents where  in   Asterisk
• except some number translation, everything worked till I migrate all users to CUCM. 

Sip integration was straight forward. But you may need to do some fine tuning. 

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[gt364555]

Thanks NithinElu! As you mentioned our setup is kind of reverse, the difference being in our setup, CUCM is used to both forward incoming call from a particular route to Asterisk and later again to handle call forwarded from Asterisk on the inbound SIP trunk. Hopefully, as your setup, this also should work without problems.