topic Re: Voip line gone mute in Tech Help - Software/Hardware etc

Voip line gone mute

softhedgehog — Thu, 20 Nov 2025 15:38:20 GMT

In order to experiment before I am forced to lose my landline, I bought a LinkSys PAP2 phone adapter and bought a new number from A&A. I can plug in a space (old-style) phone into this to make calls on this number.

This has been successful so far, but today I can hear the person on the other end, but they cannot hear me.

A&A's help pages suggest it might be a NAT problem, but the suggestions they give are either inapplicable (e.g. switching off ALG) or don't appear to make any difference (e.g disabling UPnP).

Does anyone have any suggestions for allowing Voip through the Hub2 router?

Re: Voip line gone mute

MisterW — Thu, 20 Nov 2025 15:54:17 GMT

@softhedgehog whilst I'm not familiar with the PAP2 , it does sound like a NAT problem.

There's no problem using the Hub 2 with a voip system.

First make sure that the SIP Alg is disabled on the Hub 2 g( it is by default )

Then make sure that NAT keepalive is configued on the PAP2 , google 'linksys pap2 nat keepalive' for the details

Set the NAT mapping Enable to Yes, and set NAT Keepalive Enable to yes, and the NAT Keepalive interval to say 15 seconds

HTH

Re: Voip line gone mute

PhilipHeyes — Thu, 20 Nov 2025 16:25:36 GMT

We have a Cisco SPA112 a similar ATA.    Sounds like your configuration is close.

One way voice issues can be a NAT issue, so make sure you have a STUN server configured:

E.g: stun.aa.net.uk

If present SIP ALG on the Router tends to affect in bound voice data on the default SIP Port 5060,
options are disable ALG on the router or use a non-standard ports.

We are using SIP Port 47160   and RTP Port Min 47104 with RTP Port Max 47120

Re: Voip line gone mute

MisterW — Thu, 20 Nov 2025 17:04:38 GMT

so make sure you have a STUN server configured:

I dont believe A & A provide a STUN server. Their voice servers are NAT aware and so providing the NAT pinhole is kept open ( using NAT keepalive ) then neither STUN or a SIP ALG is needed.

The A & A config for the PAP and SPA is here https://support.aa.net.uk/VoIP_Phones_-_Cisco_SPA

Re: Voip line gone mute

rogerwhill_foru — Thu, 20 Nov 2025 17:19:11 GMT

I tried to get the LinkSys PAP2 phone adapter working, but I couldn't remember what I had changed. It is possible to enter a code via the handset to reset to the default settings, change a few settings (as specified by A&A, https://support.aa.net.uk/VoIP_Phones_-_Linksys_PAP2 ).

I think I used these instructions.

----------------------------------------------------------------------------------------------

To reset the ATA device to the factory defaults, perform the following steps:
1. Connect an analog phone to the ATA device and access the IVR by pressing the
asterisk key four times: ****
Press the appropriate code to reset the unit:
• Press 877778# to reset the unit to the defaults as it shipped from the ITSP.
This will reset the User account password to the default of blank.
• Press 73738# to perform a full reset of unit to the factory default settings.
The Admin account password will be reset to the default of blank.
2. Press 1 to confirm the operation.
Press * to cancel the operation.
3. Log in to the unit using the User or Admin account without a password and
reconfigure the unit as necessary.

-----------------------------------------------------------------------------------------------------

I thought that the LinkSys PAP2 ran too warm, and bought a Grandstream 802 v2.
https://www.grandstream.uk/product/grandstream-ht802-v2-2-port-fxs-ata-telephone-adapter/

I reset it a few times while setting it up...

I didn't need to use NAT

Re: Voip line gone mute

softhedgehog — Thu, 20 Nov 2025 17:45:04 GMT

Thanks to all who have replied.

A&A do provide a stun server (stun.aa.net.uk) which I am already using.

I am not *intentionally* using NAT. I didn't even know I was. Can I NOT use it?

Re: Voip line gone mute

MisterW — Thu, 20 Nov 2025 17:56:28 GMT

I am not *intentionally* using NAT. I didn't even know I was.

Your router will be using NAT

Can I NOT use it?

Re: Voip line gone mute

rogerwhill_foru — Thu, 20 Nov 2025 18:39:59 GMT

Sorry, I was mixing up terms.

I didn't use STUN

Re: Voip line gone mute

MisterW — Thu, 20 Nov 2025 18:45:02 GMT

Then just make the settings as I posted in post 2

Re: Voip line gone mute

MisterW — Fri, 21 Nov 2025 08:01:49 GMT

@softhedgehog

There's a really good explanation of how SIP works with NAT here https://voipstudio.com/blog/sip-nat-traversal/

When you say you have one-way audio, does the call start initially with two-way and then fail to one-way after a few seconds ?

Check the Use Symmetric RTP setting in the PAP2

Re: Voip line gone mute

PhilipHeyes — Fri, 21 Nov 2025 08:19:31 GMT

Usually a valid configuration keeps working until a change or fault occurs.

So, have you tested the telephone in question to ensure that its microphone is still working on a BT phone socket ?

Sometimes the too daft to consider can be the problem !

Re: Voip line gone mute

softhedgehog — Fri, 21 Nov 2025 12:17:29 GMT

The telephone works fine in the normal land-line socket.

There is just no sound transmitted from the start.

I had not changed anything between the last use and this, hence my puzzlement.

I'll try resetting.

Re: Voip line gone mute

PhilipHeyes — Fri, 21 Nov 2025 12:46:26 GMT

MicroSIP Lite is a handy VoIP phone for MS Windows.

By comparison to a Linksys / Cisco ATA it is very quick and simple to setup
and a handy way to confirm your SIP configuration info works.

https://www.microsip.org/downloads

Re: Voip line gone mute

softhedgehog — Fri, 21 Nov 2025 12:52:07 GMT

I can't see anything about Symmetric RTP

The only use of the word "Symmetric" is labelled "FAX Codec Symmetric" (set to "Yes")

There is a RTP section on the SIP tab which does ports and a few other things

Re: Voip line gone mute

PhilipHeyes — Fri, 21 Nov 2025 13:03:41 GMT

Attached is a screen shot of our SPA112's RTP and NAT settings.

The RTP ports are important as these are accepting the in bound voice traffic during a call.

Happy to share any other menu items if needed.

Re: Voip line gone mute

MisterW — Fri, 21 Nov 2025 13:39:08 GMT

I can't see anything about Symmetric RTP

In that case , I'll leave Philip to sort you out with using STUN. Myself, I've never needed to use it...

Re: Voip line gone mute

softhedgehog — Fri, 21 Nov 2025 15:07:39 GMT

@PhilipHeyes wrote:

Attached is a screen shot of our SPA112's RTP and NAT settings.

The RTP ports are important as these are accepting the in bound voice traffic during a call.

My settings look the same as ours, except that the port min/max are 16384/16482

Re: Voip line gone mute

softhedgehog — Fri, 21 Nov 2025 16:16:23 GMT

I reset and made the necessary changes to make it connect again. Still no luck. I can hear the other person, but they can't hear me.

Re: Voip line gone mute

PhilipHeyes — Fri, 21 Nov 2025 16:19:14 GMT

Setup MicroSIP on a PC or laptop with the same type of SIP configuration and see if that gets 2 way voice.

Re: Voip line gone mute

softhedgehog — Sat, 22 Nov 2025 08:32:02 GMT

An incoming call seemed fine. Outgoing still muted. Might be random, of course.