No Audio from IVR when Calling from One Number/System

I’m experiencing an interesting issue that just started recently over the last few months. I have a PBXact setup for a customer (property management co) at their office (we will call this system A). At my office, I have a FreePBX system on premise for a non-profit that I run (we will call this system B). I also have a FreePBX setup in the cloud for my IT business (we will call this system C). Additionally, I have several FreePBX systems in the cloud for other clients.

The issue I am experiencing is that when I call system A, the property management co, from system B, the PBX will answer but I will hear nothing. Normally it would go to their IVR. But I hear nothing. I can see the call come in from the CLI in asterisk. But again, I hear nothing.

I can call system A from ANY other of the phone system’s I have out there, or my cell phone, and all works perfectly. But any time I call from system A, it connects and all I hear is silence.

I have checked to see if my CID is blacklisted in system A, but it is not. I used to be able to call to system A from system B because that’s what I would use to run call tests for the IVR, etc.

I can’t seem to figure out why, with just that one number or system, it does that. Any ideas of what could be causing this?

There are many possibilities, so you will probably have to use some tools to see what is going wrong.

I assume that’s a typo and you meant “any time I call from system B”. If not, provide details.
I am also assuming SIP trunks on both A and B. If using POTS, PRI, etc., provide details.

It is very common for trunking providers to treat on-net calls differently. Is the outbound leg on B using the same provider as the inbound leg on A? If B has multiple providers, does forcing a different outbound trunk avoid the problem? If A has multiple providers, does calling a DID on A that uses another trunk avoid the problem?

Just guessing here, maybe one of the public IP addresses changed. Confirm that in Asterisk SIP Settings, External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config, you must restart Asterisk.

At the Asterisk command prompt (on both A and B), type
pjsip set logger on
(or if you are still using chan_sip trunks for some reason)
sip set debug on
and make a failing test call. The Asterisk logs will now have SIP traces, in addition to the normal entries. If needed, use tcpdump and view the capture file in Wireshark to look at the actual RTP. If B is receiving RTP, find out why it’s not playing (wrong codec, wrong port, etc.) If not, is A sending RTP to the correct address and port? If so, if a provider media server is in the path, though unlikely, it could be a provider issue. Otherwise, I’d suspect a configuration issue with the router/firewall at one end or the other. Confirm that any SIP ALG is turned off and that the RTP port range is forwarded to the LAN address of the PBX.

If you still have trouble, paste the Asterisk logs for a failing call at and post the links here. Also, post make/model of the router/firewalls involved, as well as any VoIP-specific settings.

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