Unable to Make/Receive Calls

I’m currently running izPBX (FreePBX 16.0.19 + Asterisk 18.11.1) in a Docker container. I’m using VoIP.ms for the SIP trunk. I am unable to successfully make or receive calls.

When attempting outbound calls (extension to external numbers), the phone for the outside number rings. However, upon answering, nothing can be heard by either call participant. The call drops 5-7 seconds later.

When attempting inbound calls (external number to extension), the IVR prompt is red out. Afterward, the call drops 7-10 seconds later as well. No chance to leave a voicemail. That happens regardless of any action(s) I take, short of hanging up early.

When attempting to dial for voicemail access (*97), no automated prompts can be heard. The call, once again, drops 7-10 seconds later - regardless of any action(s) I take. So, no chance to properly setup voicemail for that extension yet.

For all three scenarios, call audio is absent. My testing was performed with a Google Voice number that I use regularly. I have it setup so that incoming callers not in my address book are prompted to read their name aloud for caller ID verification. When I call the Google Voice number from my PBX extension, call audio was absent on extension side. But on Google Voice’s side, the caller verification prompt ran - the verification audio was silent. After allowing the call to go through, the call drops seconds later.

I used the following guides to configure the instance:

  • wiki.voip.ms/article/FreePBX_(PJSIP)

If you require any info, please let me know. I will do my best to retrieve it. My configuration is probably wrong, since this is my first time working with a PBX.

You should search the posts here, Docker is not considered a good match for Asterisk due to it’s need to bidirectionally forward 10000 UDP ports to/from the container to the host by default, limit your range to just a few and you might be lucky

How many ports would you suggest to be the minimum? 100-300? 500-700?

Personally I would suggest you shouldn’t do it at all, but each call needs 4.


My instance’s compose file has this line in it:


Seems that I’ve only forwarded 100 ports so far, but my math could be suspect.

On a different note, I checked the Asterisk Log Files and saw a large number of lines that red like this:

WARNING[123] filename.c: Context 'some-pbx-component' tries to include nonexistent context 'different-pbx-component'

Something seems pretty broken in my setup. Wishing I could tell what was wrong.


read it and argue further if you care to.
Generally we disagree with your method., so don’t expect constructive critique.

And yes your math is suspect 18000-18100 is 101 ports which would cause at least 2% of your calls to fail

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