Make sure the Grandstream RTP port ranges line up with your Settings->Asterisk 
SIP Settings ranges. 5004 sounds device specific. The default for FreePBX, is, 
I think, 10000-20000. Are you using OSS Endpoint Manager? If so, see if the 
template has an RTP range setting as well.  If you are only seeing one 
endpoint’s IP address in the RTP debug, then it sounds like the other endpoint 
isn’t finding something mutual agreeable. I HATE RTP issues. SIP always seems 
fairly straight forward.  FWIW, I’m all PJSIP on FreePBX 15 (even trunks). It 
is where they’re going so I just bit the bullet and hammered it out. Once up 
and going, it does seem to be fairly robust. 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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