SIP Trunk between FreePBX and Cisco call Manager

Hello everyone,
I have created a FreePBX-Cisco Call Manager trunk. The FreePBX users can call each other without any problem. However, when I call a call manager user, the call manager user hears the FreePBX user very well but the FreePBX user does not hear the Cisco call manager user.
Please give me a hand in resolving this situation.

One way audio is normally a firewall or NAT problem. There isn’t enough information to go much beyond that.

The only thing that is marginally unusual about CUCM is that media can be exchanged with an MTP at a different address from that use for signalling, and that CUCM likes to use later offer SDP (which Asterisk can handle).

Assuming that CM and FreePBX are on LAN segments with non-NAT routing between them, confirm that in Asterisk SIP Settings, Local Networks includes the CM’s LAN. If you change this, after Submit and Apply Config you must restart Asterisk.

If the above is not applicable or doesn’t help, please paste the Asterisk log for a failing call to CM (including SIP trace) at and post the link here. If you are too new to post links, just post the last eight hex characters of the URL. Also, describe the networking between the PBXes.

This definitely also needs to include the LAN for any MTPs used. It’s a long time since I worked with CUCM, but I have a feeling that, if isn’t using MTPs, you will need the LANs of the Cisco phones, and any trunk gateway. I am pretty sure that, if not using MTPs, it uses direct media, from the start, on its side.

Thank you all for your assistance. Thanks to your recommendations, I found the solution by adding the subnet (VLAN) of the Cisco Call Manager in the local networks of the NAT settings and it works.
Thank you for everything.

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