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davicval

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Everything posted by davicval

  1. Hello, I am getting an error when trying to do a provisionning Aastra 6755i SIP phone. I have the message "No Service" on the phone display and I can't make calls ("Call failed" message when trying to dial). I saw on the PBX logs that there is an error 404: SIP/2.0 404 Not Found Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK7c403344063151107;rport=5060;received=X.X.X.X From: <sip:105@X.X.X.X:5060>;tag=e493c84887 To: <sip:105@X.X.X.X:5060>;tag=056aef94b8 Call-ID: 66e2f3184e258c58 CSeq: 17040 REGISTER Content-Length: 0 That message means: The server has definitive information that the user does not exist at the domain specified in the Request-URI. This status is also returned if the domain in the Request-URI does not match any of the domains handled by the recipient of the request. But I'm sure about the user and the password I entered so I don't see what else it could be. If someone can help me, I'll be very grateful. Thanks, David
  2. Hi, Thanks for the audio file. I tried it but apparently it doesn't change anything. But I think I know what is my problem. I created an IVR node but this one is not assigned to a specific extension or a hunt group. So my question is how can I do that? Because I have 2 extensions for my test and a hunt group but there is no link between them and the IVR node. And I don't find a field in the IVR node setup that can let me do the relation. Best regards, David
  3. Hi, I sent you an PM on 31 December with the attached wave file as you asked me. If you can tell me if the file is in the correct format, and if you know what could be the reason of my problem, that will really help me. Best regards, David
  4. Hi, I'm trying to create an IVR but I can't find the way to do it correctly. My first issue is that there is no song when I try to call the number added on the PBX. It's like I don't have an IVR node configurated. This is the configuration I did, with the filled ANI field. I paid attention to choose the right type for my wav file but after I save my configuration, everytime I try to edit the IVR node, the audio file is missing. Is it something normal? Did I miss a parameter? David
  5. 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
  6. 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
  7. 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?
  8. 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
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