Well, PJSIP can use a direct phone-to-phone path on port 5060 to transport
audio. if configured to do that. Can you change the PBX to Chan_SIP from
Chan_PJSIP for the extensions and see if your audio issues remain the same?
SIP passes audio through the PBX on 5060, so it might work differently for
you.
Aside from that, can you look in the Asterisk Info page under "Peers," do
you show your 4 phones on port 5060 listed as Avail? Here is one of mine
that is a PJSIP on a x64 FreeBBX 15 box:
Endpoint: 5232/5232 Not in
use 0 of inf
InAuth: 5232-auth/5232
Aor: 5232 1
Contact: 5232/sip:5232@10.10.150.83:5060 ff53c2b228 Avail
14.568
Gibson Prichard
Nashville, TN
gib...@prichard.tv
On Sat, Nov 13, 2021 at 10:19 PM Curt Lundgren <verif...@gmail.com> wrote:
> 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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