Oh yeah, with FreePBX 15 (at least their distro ISO), PJSIP was default on 
5060, and CHAN_SIP was moved to 5160. So, you do have to change this port # on 
your endpoints if you want CHAN_SIP out-of-the-box. Sent from  
<https://go.microsoft.com/fwlink/?LinkId=550986> Mail for Windows From:  
<mailto:verif...@gmail.com> Curt Lundgren
 Sent:  Sunday, November 14, 2021 12:34 PM
 To:  <mailto:nlug-talk@googlegroups.com> NLUG
 Subject:  Re: [nlug] Strange Asterisk issueThanks for the comments.  I've 
tried changing PJSIP to Chan_SIP; the phone
wouldn't register:

[image: old_sip.jpg]

The top status is one of the extensions registered as PJSIP, the bottom is
one after switching to Chan_SIP - it won't register.

If I turn rtp debug on, I get a series of:

Sent RTP packet to      192.168.200.40:5004 (type 00, seq 019554, ts
085649, len 000160)


Nothing else is reported.  The mystery continues.

On Sun, Nov 14, 2021 at 11:41 AM Mark J. Bailey <m...@jobsoft.com> wrote:

> 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 Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for
> Windows
>
>
>
> *From: *Curt Lundgren <verif...@gmail.com>
> *Sent: *Saturday, November 13, 2021 10:19 PM
> *To: *NLUG <nlug-talk@googlegroups.com>
> *Subject: *[nlug] Strange Asterisk issue
>
>
>
> I 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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