Remote Extension - Suddenly no audio

[MrGeezer]

Hi there,

I have a 3cx server installed in my office and at home I have provisioned a grandstream GXP-1620 using STUN as a remote extension. I set it up around 1 week ago and it was working fine. Two days ago it worked fine in the morning, then suddenly I received an incoming call and could not hear the caller. Upon testing with mobiles etc, I now find that I cannot hear any audio, nor can the other party hear me - even though the call connects fine.

This extends to phonecalls made to the other telephones within my office that are connected to the LAN, and also to phonecalls made to my voicemail - call connects but no audio.

I have done a factory reset on the remote extension and re-entered the provisioning link - but the problem persists.

Any idea what could have caused this to happen when previously working ok? Something to do with firewall ports??

 

[Saqqara]

Have you tried ticking 'PBX Delivers Audio' under Extension - Options

Being stun setup, I would setup portforwarding as per info here - https://www.3cx.com/community/threa...etgear-nighthawk-x4s-d7800.54620/#post-225296

 

[JohnS_3CX]

Hi @MrGeezer

As @Saqqara suggests, you will need to ensure the ports are open.

At home I imagine you are using a modem supplied by your ISP, which decides whether the traffic should pass through or not. If you rely on it working automatically you may find out that it is unreliable in the long run if unless you port forward manually. When done, dial *777 echo test to validate.

 

[MrGeezer]

Thanks for your messages. I already tried switching on PBX delivers audio before I reset and re-provisioned. I have now switched it OFF again but still no joy.

Under the provisioning tab of the phone it states that the local SIP port of the phone is 5065. RTP Audio starts is 14000 and RTP Ends is 14019.

I am indeed using the ISP's router (virgin super hub). I have checked the firewall and I have the folowing ports being forwarded to the internal IP of my phone:

TCP&UDP - 5065
UDP - 14000 - 14019

Still no audio

 

[JohnS_3CX]

Is there a SIP-ALG setting you can switch off on the router? Also note that some ISP router/modems might cause interference in the operation of VoIP calls in case they have lines provisioned on them already. You can also try rebooting the phone just in case (so it will start a new session).

If this does not help, run a capture on your PBX while the call is running to ensure a common codec has been negotiated and to see if there is an audio stream at all. The capture can also be done on the phone side if you wish to compare the captures between them and ensure they agree with each other.

 

[MrGeezer]

Thank you. I've done a bit of gogling and not only does my router not support disabling sip-alg, there appear to be some complaints about it generally not playing nicely with Voip. I suspect I will have to put it in modem-only config and buy another router - will do so and update thread in due course.

 

[JohnS_3CX]

Yep, that's one way to go but you may also consider using the SBC with encryption enabled to bypass the router's filtering. You can install it on a VM, a Raspberry Pi or a PC.

 

[MrGeezer]

Thanks was toying with the idea of SBC. I guess I will install it on a virtual machine on my laptop and see if it's worth paying out for a raspberry pi or similar.