No Audio on calls?

[splunge]

HI,
Have setup 3CX with 2 VOIP lines and 2 extensions. Have had difficulties with various things, mostly ports I think.

Finally got 2 VOIP lines to register Sipgate (although I believe I will have problems with them from reading these forums) and Voipcheap.

Forwarded each one to an extension and got them to ring, but when I answer there is no audio? Either from the phone I call on or the softphone I set the extension up on.

Any tips guys, I like the look of this software and seems a little easier than something like Asterisk.

Thanks in advance.

 

[N_G]

No audio would point to a firewall problem - either on your firewall or high ports being blocked by your ISP. Ensure you have reviewed the documentation on what ports to open and if you are sure all ports are open check with a packet sniffer that traffic is indeed arriving at the end point. Also use support VOIP providers and phones.

 

[Anonymous]

Yep sounds like ports.

Typically you need ports:
UDP 5060 (SIP)
UDP 10000 - 20000 (RTP)
UDP 3478 (STUN)

If that does not work, make sure you use a compliant codecs (same your ISP uses).

To test this an easy way. Setup a line on 3cx and have that one always forwarding to the VM this will trigger the "leave your message.." message it might give you some idea on where to look. If you cannot hear that message start with the ports and work your way towards codecs.

PS. do a search on "cannot make calls" or on ports there is a lot been said about this topic already.

 

[splunge]

Hi Guys,
Thanks for the replies, always appreciated.

I have forwarded a huge amount of ports and tried with my machine as being in a DMZ, but still no joy

If I just use X-Lite with my VOIP provider settings it works fine.
Under 3CX I can get both my lines in green, I'm just stumped with no audio.

Everywhere I look it seems to be port related (RTP seems to be a popular one people discuss), but I'm not having any joy.....I even tried a different router lol.

Thanks anyway guys, I'll plod along until I find something out there that works.

 

[Anonymous]

Ahh DMZ that is where you trouble might be.

If you install 3cx on the internal network it works ok for sure. Why do you have 3cx on the DMZ (Oh for those not in the know DMZ = Demilitarised zone it is a Firewall thingy and DMZ will have less security for outbound in than the stuff behind the firewall) do you want to browse to 3cx from external?

Question:
3cx is on DMZ are your phones in DMZ or Internal network?
If they are internal network you will have a problem with NAT and that can be fixed with STUN or VPN.
Alternative you can put all your phones in DMZ so no NAT should play?

 

[3CXsupport]

Hi,

If you describe better your network layout we might be able to pick out some trouble areas. Can you please detail some extra info ?

 

[splunge]

Hi all,
Nice to see a forum with people that are willing to help.

I haven't got the machine in a DMZ, it's just something I tried as a last gasp effort.

One thing that may, or may not, be the problem is that I am testing this all on the same PC.

Network is basically half a dozen machines behind a Netgear router. As I said I have tried various port forwards and have had X-Lite working on it's own, but when used with 3CX I alwys get the no audio thing.

Is this becasue it's on the same machine?
I'll have another go next week I think, weekend is here now, time off lol.

Thanks again all, love the responses.

 

[Anonymous]

Hi all,
Network is basically half a dozen machines behind a Netgear router. As I said I have tried various port forwards and have had X-Lite working on it's own, but when used with 3CX I alwys get the no audio thing.

Is this becasue it's on the same machine?

Ok if all the comps and phones are behind the NetGear router you should be able to make calls.

Not having sound could be a Codecs issue aswell (see what codecs X-lite is using). I assume the phones register in 3cx and so does the provider so all is good.

Now few things go look at:
SIP port, make sure you do not use the same sip port for incomming and outgoing (this if you call your self via your Vsp).

UDP 10000 - 20000 are for you sound so they need to be enabled on your router to pass through. You can narrow these down but that means you have to look at your phone what to use aswell on your ata.

Codecs G711 is not supported (i believe) so set all the codecs to G726 or something like that.

On your phone (if possible) configure your proxy and registration servers, I noticed that sometimes it likes to have a proxy server configured. THis in your case would be the NetGear router address.

To sum it up, you should be able to make calls to and from phones in your own network without any hassle. If that does not work you have a registration type issue.

Calls to outside network (via your sip provider) need to have the holes in the firewall.
SIP
RTC
STUN (if you need it).

hope this helps.