No sound between two remote phones

[Lukas Sukdol]

Hi, am using 3cx system with about 10 local clients (Grandstream 16xx, android, iOs) with no issues for a few months. Now I tried to connect two phones remotely (let say phone A and phone B), each in different network, actually in different countries. Both clients (A,B) can make a call to our local extensions and receive calls from them. Phone A can make a call to phone B with no issues, but call from B to A is connected but w/o sound...

3cx is configured properly (OK firewall test), both remote phones were automatically provisioned. There are Mikrotik routers in both locations with SIP ALG (SIP Helpers) turned off. Any idea what should be wrong? Or where to start to debug this problem?

 

[leejor]

I'm going to assume that these are using STUN. My guess would be an issue at the extension A end. Might be the router there, or a conflict with another device. The fact that calls work correctly when placed from local extensions suggests it is something caused by B attempting to connect directly with A. You may want to check the 3CX Activity Log to see if the registration from A looks as it should. Compare with that of B, and compare router settings at both location, what is different?

You might want to try enabling PBX delivers audio, on extension A, to see if that helps.

 

[Lukas Sukdol]

Thanks leeror, but no luck with "PBX delivers audio" option...
Both phones (extensions) seems to be registered properly:
Extension 66 is registered, contact: sip:66@my_ip:5060

And no failures in Activity log..

Note that all phones (A,B) and 3CX are behind NAT - all networks have
static IP addresses - do I need to setup NAT on client (phone) site for 5060 port?

 

[leejor]

What happens if you change both extensions to use PBX Delivers Audio? Port forwarding of the SIP port (5060) is usually not necessary if there are a limited number of SIP devices, at one location, using STUN. You say that calls are ringing, which means messages to that port are passing correctly. What does not seem to be passed are the audio ports, but only when called fron the "non local" extension. Which begs the question...why are the audio ports being passed when a local extension calls, but not when the other remote extension calls. What is different. sending that call back through the PBX, which should happen, if you set the calling Extension as PBX delivers Audio will probably fix it. It still leaves the "why" unanswered.

Extension 66 is registered, contact: sip:66@my_ip:5060

Is that the called party, or the originator? Why is it not registering with the public IP of the remote router? I would think that it should if the set is using STUN. Are there any other SIP devices at the destination?

 

[cobaltit]

@Lukas Sukdol

If you look at the provisioning tab for the phones using STUN you will see a SIP port and RTP port range. Those are supposed to be forwarded to the remote clients at their respective firewalls. You can usually get away without doing it but it's the first thing 3CX support will tell you to do.

 

[Lukas Sukdol]

@leejor
"PBX delivers Audio" did not resolve problem (tried on A site and also on both extensions).

Both phones are properly registered with theirs public_IP - it means there are 3 public static IPs,
one on phone A site, second on phone B site, third on PBX site. On the provisioning tabs there are same values for Local SIP Port of phone and RTP Ports (5060, 14000-14019 resp.) - but there is only one SIP device in A and B networks, so I suppose its no problem.

@cobaltit
On the A site I tried to port forward public_IP_of_A:5060 to the A phone local IP - but still no success.

 

[Justin D]

You also need to forward the RTP ports as these carry the audio.

Also, and to make it easier for you diagnose issues going forward, have you thought about using a different SIP port for each phone?

 

[Lukas Sukdol]

@Justin D
thanks - i will try it - by now I have a strange workaround - because I am on phone A, I can call everywhere, calls from phone B need to be received on mobile client (Android). Android client is on the same network (on wifi), and same extension - so both of them ring on incoming calls...

 

[leejor]

If you use identical routers, at each remote end, then either the settings on the routers differ, or it is a setting on the set itself. If , at the receiving end, you have two SIP devices, behind a router, both using STUN, then each will need a unique local port number. Even so, a call to one device, may mean that the audio ports will now always go to that device, until the router is reset, or you have forwarded certain ports to a local IP. If the second device is a 3CX App, using the tunnel, then that should not happen.

A single device, using STUN, behind a remote router, will usually not require any forwarding. Once you go beyond that, depending on the router and how you have configured the devices, you can get the "no audio" issue. It is always recommended that sites with more than one device, make use of the 3CX SBC.

 

[Sergio Fernandez]

Hi

Are you LAN ip the same range at both sites?
eg 192.168.1.X and 192.168.1.X
The default range for a lot of manufactures (eg Netgear) is 192.168.1.X

Is so make one network 192.168.10.X with DHCP pushing out the same range (eg 192.168.10.10 to 192.168.10.100) Range can be made bigger or smaller as required.
Be ware that other key item on network will need to have their IP address changed if they are set manually.

Hope this offers an idea.