No audio between 2 calls

[davicval]

Hi everyone,

 

I'm trying to use Vodia PBX on one of my Ubuntu server hosted at OVH.

 

Installation was ok, the trunk SIP given by OVH was ok too and I created two extensions for phones that I have at work. Everything was working for the best.

 

But yesterday, my company wanted to add a firewall (Fireware XTM WatchGuard).

So now I have a private IP address for my PBX and all traffic is going to the firewall which redirects everything, as you can imagine.

 

 

I opened all the ports needed, I think at least.

The biggest problem I have now is that there is no sound when I try to call someone.

 

I tried to open RTP ports configured in the PBX, to see if it can be another hidden option but I got nothing.

 

Do you have any solution I could try? After spending so much time on it, any idea will be really appreciated.

 

Sorry for my english if you see something wrong.

 

 

Cheers,

David

[Vodia support]

Hi David

 

You're best bet would be to use the "IP Routing List" and "SIP IP Replacement List"

check the article below.

 

http://vodia.com/documentation/admin_sip_and_audio

[davicval]

Hi,

 

Thanks you very much for the link, the parameter IP Routing List was the problem. Just full in it and I can hear sound now.

 

I just have one problem left now before I could consider using Vodia PBX for my company.

 

When I try to call the number given to the trunk SIP, a message tells me "the person is not available" (I'm French so the message could be a little different in English).

 

The parameter "Destination for incoming calls" in Trunk Settings is by default set to "Send all calls to the Request-URI destination".

But when I try to change this parameter to "Send all calls to a specific account", everything is working.

 

What I want is to call the number of the trunk SIP and hear all the phones provisionned ringing.

 

Do you know if there is a specific parameter for that or is it a problem of the trunk SIP provider?

[Vodia PBX]

In France it is the easiest to "match extension after prefix" for inbound trunk routing. For example if your company has the number +3312345xx (xx being the extension, for example 40, 41, 42 but also 0), then you can use the prefix 12345 or 2345. it is really just about matching something in the number and then taking the rest to determine the destination in the PBX.

[davicval]

I agree with you, it could be useful to configure it like you said. But in this case, you have to pay lines for each extension you have to add. I mean if the number given to the trunk SIP is +331234500 and I want to have 2 extensions (01 and 02 for example), I will have to pay to have two more lines to have +331234501 and +331234502. Am I right?

 

However, we are using a different working in my company. Let me explain how we work:

When someone call us, there is an audio message telling to choose between 1 or 2 to get the commercial office or the technical office. If someone choose 1, just the phone at the commercial office will ring.

 

But if someone choose 2, the phones located in the two offices will ring. And that's the part I really not understand where to configure it.

 

Tell me in you need more informations or if something is not clear about what I said.

 

 

Cheers,

David

[davicval]

Ok, I can answer to myself, I have just found what I really wanted to do.

 

I needed to create a hunt group and add the extensions I wanted. It's just that the paramater "hunt group" translation in French is not the best but anyway, everything is working now.

 

Thanks again for both answers, they really helped me.

 

 

Cheers,

David