It's all working now, but as with so many things, it's unclear just which
tweak fixed what problem.  Turning off RTP checksums and Strict RTP got the
extensions working with each other, a tweak with my SIP provider did the
rest.

I very much appreciate the responses, tips and hints - they really did get
me looking in the right places.  Thanks again!

On Sun, Nov 14, 2021 at 12:46 PM Mark J. Bailey <[email protected]> wrote:

> 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 Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for
> Windows
>
>
>
> *From: *Curt Lundgren <[email protected]>
> *Sent: *Sunday, November 14, 2021 12:34 PM
> *To: *NLUG <[email protected]>
> *Subject: *Re: [nlug] Strange Asterisk issue
>
>
>
> Thanks 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      *MailScanner warning: numerical links are often
> malicious:* 192.168.200.40:5004 <http://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 <[email protected]> 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 <[email protected]>
>> *Sent: *Saturday, November 13, 2021 10:19 PM
>> *To: *NLUG <[email protected]>
>> *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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