rtp set debug on And make an internal call between extensions. Observe the IP 
and negotiated RTP port#s for the 2-way, 2-channel audio between the two 
endpoints (internal or mixed). Audio issues are almost always related to some 
setting that is causing this negotiation to not be mutual agreeable.  Be sure 
to ‘rtp set debug off’ when done as it will continue displaying (and logging) 
these RTP traces, and any calls of substantial length will generate thousands 
of messages each (even when not in the asterisk CMD prompt mode). Sent from  
<https://go.microsoft.com/fwlink/?LinkId=550986> Mail for Windows From:  
<mailto:verif...@gmail.com> Curt Lundgren
 Sent:  Saturday, November 13, 2021 10:19 PM
 To:  <mailto:nlug-talk@googlegroups.com> NLUG
 Subject:  [nlug] Strange Asterisk issueI just spun up the latest version of 
RasPBX, Asterisk on the Raspberry Pi.
Asterisk is version 16.13.0, and FreePBX is 15.0.17.55.

There are four Grandstream VOIP phones as extensions, all registered as
PJSIP, and there's one SIP trunk going out to the world.  The PBX and the
phones are on the same subnet.

All of the phones can receive audio, but none can send audio to another
device.

   1. Call the speaking clock at *60, it works just fine, Allison's voice
   is heard.
   2. Call from one extension to another, no audio.  They're probably both
   receiving audio, but neither can send any.
   3. Make a call outgoing on the SIP trunk, the local phone can hear the
   other party but cannot be heard.
   4. Receive a call from the SIP trunk - again, the phones can hear the
   other party, but cannot be heard.

I don't think it can be any kind of NAT issue, since the phone extensions
and the PBX are on the same subnet.  Aside from the phones not sending
audio, everything seems to be working.  If I bring up Asterisk Info in
FreePBX it shows the extensions and the trunk using PJSIP with ONLINE
status.

It's got to be something simple and dolt-ish I'm doing or have failed to
do.  Any ideas?

Thanks,
Curt Lundgren

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