VPN one way audio.

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Roy Reynolds

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Jun 24, 2025, 11:57:57 PMJun 24
to sipxcom-users
Hello, 
I'm having a frustrating issue where remote users connect via VPN and register with the sipxecs pbx. The remote user can initiate a call and ask parties have audio.  If a user on the same subnet as the pbx makes a call from their desk phone, there is one way audio.

I have put the vpn subnet in the internet calling subnet and used /8 as the mask.

Any ideas for what to check? 

Thanks in advance.


OnRelay Support

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Jun 25, 2025, 3:50:03 AMJun 25
to sipxcom-users
Ensure SIP ALG is disabled on any router in the path, and / or better, secure the traffic with TLS and SRTP, which prevents any such application level router mechanism from modifying SIP packets.

Michael Picher

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Jun 25, 2025, 6:49:28 AMJun 25
to OnRelay Support, sipxcom-users
If it's not some SIP ALG, it could be a routing issue. The phone needs to have a default gateway that knows how to route to the VPN's IP range. 

If you're on a PC that has the same IP parameters as a phone, can you ping a VPN'd in IP address?

Mike
--------------------------------------------------------
Michael W. Picher


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Roy Reynolds

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Jun 25, 2025, 7:09:14 AMJun 25
to Michael Picher, OnRelay Support, sipxcom-users
Mike, 

I'll try to ping the remote device from a pc on the same subnet as the pbx and see what happens. I have the same issue at more than one location and I know sip alg is off so I am thinking it's a routing issue. I did have the customer add a route but didn't try the ping.  The fact that audio works perfectly if the remote vpn user initiated the call made me think that routing was okay. 

Thanks for the suggestions. 

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Roy Reynolds

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Jun 25, 2025, 9:04:53 AMJun 25
to Gerald Drouillard, sipxcom-users
Here you go: 

"Alias"<sip:Ex...@domain.com> "Alias"<sip:Extn@vpnipaddre:46843;transport=udp;x-sipX-nonat>

I changed some information to protect the customer so: 
Alias is the person's name
Extn is the extension number
vpnipaddre is the remote vpn ip.

I do notice a port of 46843 but not sure why that is there.

Mike,

I can ping the remote ip address of the registered client without an issue so it doesn't appear to be a routing problem. 

The signaling works but rtp is not finding a route for some reason if the call is initiated from the pbx subnet.


On Wed, Jun 25, 2025, 7:36 AM Gerald Drouillard <ger...@drouillard.ca> wrote:
show us a few of the registrations of a vpn extension in sipxecs

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/CALz8TxmreMiEfjk4VPnSEO-aceGWCjZjmZ2WGrLz9dBSV1RPiw%40mail.gmail.com.

Michael Picher

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Jun 25, 2025, 9:08:20 AMJun 25
to Roy Reynolds, Gerald Drouillard, sipxcom-users
I think I know what it is...  I think you need to add your other subnets to the NAT Traversal settings.

Mike
--------------------------------------------------------
Michael W. Picher

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/CALz8TxkOxn1Xn0MYmtDYBd__QOXcW4kL6QKw1T7QyWZLg6_6nw%40mail.gmail.com.

Roy Reynolds

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Jun 25, 2025, 9:11:03 AMJun 25
to Michael Picher, Gerald Drouillard, sipxcom-users
I assume this is under Internet calling? If yes, I already did that. If no, then I don't know where to add those subnets. 

Michael Picher

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Jun 25, 2025, 9:41:06 AMJun 25
to Roy Reynolds, Gerald Drouillard, sipxcom-users
Dang, thought for sure this was it.....
--------------------------------------------------------
Michael W. Picher

Jhony Perez

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Jun 25, 2025, 9:49:30 AMJun 25
to Roy Reynolds, Michael Picher, Gerald Drouillard, sipxcom-users
I'd normally start with just calling the voice mail and leaving a message, that way you can verify two way calling between the phone and the PBX, if you can hear the greeting and the message have audio, then you know your VPN to the system is good, then move to a call to a local ext. then finally an external call. Once you identify that, then we can troubleshoot that issue.

Thanks,
Jhony

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/CALz8Tx%3D0fO9ePP9syXada3F%2Bn1CgRNtYDiW5VYqLRBTmyhoQEQ%40mail.gmail.com.

Gerald Drouillard

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Jun 25, 2025, 12:22:19 PMJun 25
to Roy Reynolds, sipxcom-users
Is your VPN IP's in the Intranet Domains?
The next thing would be to look at a packet capture in wireshark and see what is going on.

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/CALz8TxkOxn1Xn0MYmtDYBd__QOXcW4kL6QKw1T7QyWZLg6_6nw%40mail.gmail.com.

Nathaniel Watkins

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Jun 25, 2025, 1:45:43 PMJun 25
to Gerald Drouillard, Roy Reynolds, sipxcom-users
Install sngrep on the server.

Run sngrep -c from cli.

Make the call. View SIP trace.

Roy Reynolds

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Jun 25, 2025, 3:08:20 PMJun 25
to Gerald Drouillard, sipxcom-users
Yes. I put them in.  I did a tcpdump and will analyze it soon. I tested a call myself whilst on the vpn and could hear them when they call me but they could not hear me.

When I called we all heard each other. 


Donkiss Boss

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Aug 21, 2025, 6:11:30 AMAug 21
to sipxcom-users
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