Strange Asterisk issue

[Curt Lundgren]

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

[Gibson Prichard]

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:52...@10.10.150.83:5060            ff53c2b228 Avail        14.568

Gibson Prichard

Nashville, TN

gib...@prichard.tv

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[Mark J. Bailey]

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 for Windows

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[Curt Lundgren]

Thanks for the comments.  I've tried changing PJSIP to Chan_SIP; the phone wouldn't register:

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.

To view this discussion on the web visit https://groups.google.com/d/msgid/nlug-talk/1062413990.166242.1636911661843.JavaMail.zimbra%40mail.jobsoft.net.

[Mark J. Bailey]

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 Mail for Windows

 

From: Curt Lundgren
Sent: Sunday, November 14, 2021 12:34 PM
To: NLUG
Subject: Re: [nlug] Strange Asterisk issue

 

Thanks for the comments.  I've tried changing PJSIP to Chan_SIP; the phone wouldn't register:

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 (type 00, seq 019554, ts 085649, len 000160)

 

To view this discussion on the web visit https://groups.google.com/d/msgid/nlug-talk/CADPPtroeqmw2ywEn0k7wqLnOWdS0f%2Bf-xGPP0nCZ01M0vRxKXQ%40mail.gmail.com.

[Mark J. Bailey]

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 for Windows

 

From: Curt Lundgren
Sent: Sunday, November 14, 2021 12:34 PM
To: NLUG
Subject: Re: [nlug] Strange Asterisk issue

 

Thanks for the comments.  I've tried changing PJSIP to Chan_SIP; the phone wouldn't register:

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 (type 00, seq 019554, ts 085649, len 000160)

 

To view this discussion on the web visit https://groups.google.com/d/msgid/nlug-talk/CADPPtroeqmw2ywEn0k7wqLnOWdS0f%2Bf-xGPP0nCZ01M0vRxKXQ%40mail.gmail.com.

[Curt Lundgren]

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!

To view this discussion on the web visit https://groups.google.com/d/msgid/nlug-talk/223726025.166441.1636915575419.JavaMail.zimbra%40mail.jobsoft.net.