Voip line gone mute

[#1]

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?

[#2]

@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  

[#3]

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

[#4]

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

[#5]

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

[#6]

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?

[#7]

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

Your router will be using NAT

Can I NOT use it? 

No

[#8]

Sorry, I was mixing up terms. 

I didn't use STUN

[#9]

Then just make the settings as I posted in post 2

[#10]

@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 

[#11]

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 !

[#12]

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.

[#13]

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

[#14]

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

[#15]

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.