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most_ahdy
New Contributor

FortiVoice to CUCM SIP trunk

Hi,

   I wonder if some has experience in establishing a SIP trunk between FortiVoice and Cisco Call Manager?

9 REPLIES 9
Jean-Philippe_P
Moderator
Moderator

Hello most_ahdy, 

 

Thank you for using the Community Forum. I will seek to get you an answer or help. We will reply to this thread with an update as soon as possible. 

 

Thanks, 

Jean-Philippe - Fortinet Community Team
Anthony_E
Community Manager
Community Manager

Hello,

 

We are still looking for someone to help you.

We will come back to you ASAP.

 

Anthony-Fortinet Community Team.
Anthony_E
Community Manager
Community Manager

Hello,

 

May I invite you to have a look at this document?:

https://docs.fortinet.com/document/fortivoice-enterprise/7.0.1/fortivoice-phone-system-administratio...

 

Regards,

Anthony-Fortinet Community Team.
most_ahdy
New Contributor

I had a look to this section of administration guide, but it did not work with a sip trunk, but worked only using custom office peer.

I am in a testing phase, to ensure that all are ok,

but I wonder if anyone has experience in integrating Cisco Cucm with FortiVoice, 

I need to route calls from Fortivoice specific extensions to CUCM specific PSTN gateway, so the call routing in CUCM will be based on Fortivoice caller number, any body has this scenario?

jburk
New Contributor

Are you still needing help on this? I have done this configuration several times.

JB - CCNA R&S, Wireless, Voice. CCNP Collaboration. VCP6-DCV
JB - CCNA R&S, Wireless, Voice. CCNP Collaboration. VCP6-DCV
Hamzeh
New Contributor

Hello Jburk, 

 

I'm kind of new to the voice world and I was looking for an answer to similar question, can we integrate any SIP server such as CUCM or from any other vendor with FortiVoice and is the right way to do it would be by creating a SIP trunk port between the two local SIP servers?

 

If so, is there a guide we can follow? I've been reading some guides and I didn't anything helpful. 

 

I'd truly appreciate your input on this one. 

 

 

Yip
New Contributor

Hello @jburk 

 

We have had many problems with making conferences. Validation was done with the following tests:

Host: CISCO. Aggregates: External, External. OK
Host: Fortinet. Aggregates: External, External. OK
Host: Fortinet. Aggregates: Fortinet, Fortinet. OK
Host: CISCO. Aggregates: CISCO, CISCO. OK

Host: Fortinet. Aggregates: CISCO, CISCO. OK

 

But the following ones don't work:

Host: CISCO. Aggregates: Fortinet, CISCO. NO
Host: CISCO. Aggregates Fortinet, Fortinet. NO

 

Do you have any idea what we might be doing wrong?
Regards

Yip
New Contributor

Hello!
The specific problem I reported was not solved anywhere. We checked the issue with Cisco-Avaya and found a case with the same characteristics. It is a codec issue, in our initial configuration in FortiVoice, we had the following codecs in the SIP Trunk:

set audio-codecs g711u g711a g729 g722 g723.1 g726 opus gsm

We removed the g726 opus and gsm.
set audio-codecs g711u g711a g729 g722 g723.1

This solved the problem.

I share with you the link of the Cisco-Avaya solution that helped us with Cisco-Forti.

https://community.cisco.com/t5/ip-telephony-and-phones/issues-ad-hoc-conferencing-cisco-and-avaya-ph...

 

Regards.

callmeahero
New Contributor II

Steps for SIP Trunk Setup:


Configure the SIP Trunk on FortiVoice:

Go to System > Network > SIP Trunks in the FortiVoice interface.
Click on Create New and select SIP Trunk.
Enter the IP address or FQDN of the CUCM server.
Set the Port (usually 5060 for non-encrypted SIP, or 5061 for TLS).
Configure Codec Settings to match what CUCM supports (e.g., G.711, G.729).


Configure the SIP Trunk on CUCM:

In CUCM, go to Device > Trunk.
Click Add New and select SIP Trunk.
Set the Device Name and Destination Address (the FortiVoice IP address).
Under Protocol, select SIP and set the Port (same as configured on FortiVoice).
In the SIP Trunk Configuration, set the Codec Preferences to match those on FortiVoice.


Dial Plan and Routing:

Ensure that both systems are using compatible dial plans. You'll need to configure Inbound and Outbound Dial Rules on both FortiVoice and CUCM to route calls appropriately.
For outbound calls from FortiVoice to CUCM, you may need to map the local FortiVoice extensions to a trunk on CUCM.
Similarly, for inbound calls from CUCM to FortiVoice, map the CUCM extensions or routes to FortiVoice.


Firewall and NAT:

Ensure that SIP and RTP ports (5060/5061 for SIP, and 16384-32767 for RTP) are open in your firewall.
If either system is behind NAT, make sure NAT traversal settings are configured properly on both devices.


Testing and Troubleshooting:

After setup, test the trunk by making test calls between FortiVoice and CUCM.
If calls fail, check the debug logs on both FortiVoice and CUCM for errors. Look for SIP registration issues, codec mismatches, or routing problems.

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