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External extensions not working


eukirne
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Hi everyone,

 

I've got a problem. I've Axon in a location with fixed IP and other location with Express talk, Xlite and a videophone.

 

The problem I have in the remote location is that the phones register but I can't make calls and in the Axon side if I try to call to the remote one I get "error returned: temporarily unavairable" in the moment the other side answer, no matter if I use a hardware phone or a softphone (Express talk or Xlite) the same message is returned.

 

I've configured the public IP, opened the ports, everything. I thing it could be a problem of the RTP port but I'm not sure.

 

Any idea?

 

Thanks and regards,

Eukirne

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I've got a problem. I've Axon in a location with fixed IP and other location with Express talk, Xlite and a videophone.

 

Off the top of my head:

 

1. If the remote clients are using dynamic IP's, maybe their IP changes between the time they register with Axon and the time a call comes in for them, making the hosts unreachable?

 

2. Correct me if I'm wrong, but SIP uses UDP 5060 to handle call progress (calling out, etc.), and RTP for actual voice data. If a call can't even be established, make sure UDP 5060 is routed through the firewall to the right device inside. If you have more than one SIP device in one location, make sure they are configured to each listen on a different port (eg. device #1 -> UDP 5060, device #2 -> UDP 5061, etc.) and that the firewall maps ports accordingly. Until this works, you won't get anywhere

 

3. RTP is similar: Each device and the firewall must be configured to allow one UDP port and the next one (eg. 5004 and 5005 on GrandStream IP phone; one port for RTP, the next for RTCP). But if you can't even get phones to ring, don't worry about RTP yet.

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Off the top of my head:

 

1. If the remote clients are using dynamic IP's, maybe their IP changes between the time they register with Axon and the time a call comes in for them, making the hosts unreachable?

 

2. Correct me if I'm wrong, but SIP uses UDP 5060 to handle call progress (calling out, etc.), and RTP for actual voice data. If a call can't even be established, make sure UDP 5060 is routed through the firewall to the right device inside. If you have more than one SIP device in one location, make sure they are configured to each listen on a different port (eg. device #1 -> UDP 5060, device #2 -> UDP 5061, etc.) and that the firewall maps ports accordingly. Until this works, you won't get anywhere

 

3. RTP is similar: Each device and the firewall must be configured to allow one UDP port and the next one (eg. 5004 and 5005 on GrandStream IP phone; one port for RTP, the next for RTCP). But if you can't even get phones to ring, don't worry about RTP yet.

 

 

Hi Fredtheman,

 

The IP doesn't change, I'm sure.

 

The phones rings, but the call hangs up in the moment the other side answer.

 

I'll test the ports in the router / firewall to make sure they're right. Because, as I have more than one SIP account in the same location maybe they are in the same port.

 

Thanks!! :)

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I'd made more tests, in the remote site I've got a C&S vizufon 5300 (a videophone) and I can call to a local extension and I have video but not audio (the problem with the video was I was using too much bandwidth).

 

but if I try to call from the local videophone to the remote one I still have the error message: "temporarily unavariable" I've noticed that it happens everytime I try to call to something outside the site.

 

I'd configured a FWD account as external line and I can receive calls but I can't make them.

 

I think it can be something about the local ports, but I've configured the NAT to redirect the ports to Axon PBX PC. I've redirected 5060-5090 and 8000-8100.

 

Any idea?

 

Thanks!!

 

Eukirne

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I'd made more tests, in the remote site I've got a C&S vizufon 5300 (a videophone) and I can call to a local extension and I have video but not audio (the problem with the video was I was using too much bandwidth). but if I try to call from the local videophone to the remote one I still have the error message: "temporarily unavariable" I've noticed that it happens everytime I try to call to something outside the site. Any idea?

 

Remember, I'm just as much a newbie as you are, but if you can make phones ring each other (either a phone on the Axon side to the remote side, or the remote side to the Axon side), it means that you have SIP working ok. However, if you get no sound once the calls are established, I would first guess it's an RTP issue. You have to check which ports each SIP device is supposed to use (ideally, you should force which ports it uses: For instance, the GrandStream IP phone uses 5004 for RTP - and probably 5005 for RTCP), so that you can map just that port on the NAT firewall. Do this on both sides, obviously.

 

If you still get no sound, maybe it's a codec issue. Make sure they both use G711u or G711a by default, as this is the most standard codec.

 

If you still can't get them working, go ask in VoIP-related forums like those:

 

news://comp.dcom.voice-over-ip

http://voxilla.com/PNphpBB2.html

 

HTH

Fred.

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Thanks a lot Fred!

 

I'd revised the ports and they seems to be ok.

 

I'd tried another thing. I'd configured a FWD account as an external line and I receive calls normally, including sound. But I can't make calls.

 

But in the same site (Axon side) I shut down Axon PBX and I configure a softphone with a FWD account in the same computer and I can make and receive calls normally.

 

So the problem is my Axon configuration, cause if the network wasn't working I should no be able to make calls with the FWD softphone.

 

now the problem is that I don't know what is wrong.

 

here is my Axon config:

 

settings --> network

 

- Local SIP port: 5060

- Local RTP ports starting: 8000

 

in "allow incoming calls and audio through into private networks" I'd tried with stun, and static IP with ports 5060 and 8000.

 

The line configuration is the same that it's explained in the NCH web.

 

I'm completely lost :blink: , anyone can help?

 

Thanks!

 

Regards,

Eukirne

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