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aw



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#57504 15-Feb-2010 19:42
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Hi,

I've got an issue with by VFX Asterisk box. I'm posting here as instructed by Xnet support.

I have two VFX lines - one I've had on Asterisk for ages (we'll call line 2), and one I just had moved over on Monday (we'll call line 1 as it is my original VFX line from when I signed up). "Line 1" was previously on my PAP2T box.

I've set up the trunk and routes etc etc via FreePBX and there is one thing I cannot get to work - incoming calls on line 1.

I've verified everything else works - calls both ways on Line 2, and outgoing calls on line 1. I've verified the correct CallerID shows on my cellphone so I have successfully tested using both lines for outgoing calls.

I've used Wireshark to verify SIP registration is working for both lines - it is, I get SIP/2.0 200 OK responses for both lines from WxC's server at pan.wxnz.net/58.28.20.150.

Wireshark shows no activity when I pick up my cellphone and dial my Line 1. The call goes without delay to Xnet voicemail.

The Wireshark filter I am using is "host 58.28.20.150 and udp port 5060".

I have tried both domains as.wxcnz.net as per the VFX/Asterisk instructions here on GZ and pan.wxnz.net as instructed in the provisioning e-mail with no noticeable difference. I simply cannot get incoming to work on Line 1.

Anything else I can check? Am I right in assuming that if registering is successful but if no UDP activity is seen when trying to make an incoming call to the affected line (10-30 seconds after seeing the successful register), the problem is at WxC's end? What else can I chack?

I will PM the actual phone numbers and my Xnet username if asked.

Thanks in advance for any help,

Andrew

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  #299197 15-Feb-2010 19:47
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Somebody a long time ago had this issue as well..

If you do a "sip show registry" from the asterisk CLI do you see a Registered state next to both lines?


aw



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  #299215 15-Feb-2010 20:08
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Yep - (stripped) output below:
as.wxcnz.net:5060 994XXXXX 165 Registered Mon, 15 Feb 2010 20:05:30
as.wxcnz.net:5060 995XXXXX 165 Registered Mon, 15 Feb 2010 20:05:30

Yep, that's a ported ex-Woosh number at the top. That's the one that works! It's the 95xxxxx one that's not working for incoming calls.

The 95xxxxx was up until this morning provisioned to a PAP2T box.

 
 
 
 


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  #299249 15-Feb-2010 20:31
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PM your numbers and I will take a look tomorrow ... after 8 as I don't want to wake you with a test call as I start early :),

Generally if it goes straight to VM then the system thinks there is no registration, if there was a delay before VM then it would generally be some sort of SIP signaling issue to the device.




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications

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  #299262 15-Feb-2010 20:43
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Do you have any port forwarding set up in your router?

aw



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  #299293 15-Feb-2010 21:17
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@Maverick: PM sent - thanks for the quick response! (what are you doing working so late?)

@sbiddle: Yep, UDP ports 5060, 8000, and 10000-20000 all forwarded to the asterisk box.

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  #299296 15-Feb-2010 21:21
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What are you using port 8000 for?

If you're using it instead of port 80 I'd high recommend not having the web interface public facing. If you are lock down the root dir so you need the password to access the default web page.





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  #299300 15-Feb-2010 21:26
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What sort of router do you have? Does it have any sort of SIP ALG stuff in it?

 
 
 
 


aw



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  #299342 15-Feb-2010 22:26
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Hi, It's a Dynalink RTA1025W and yes there is a SIP ALG on this router and it's enabled.

That's just UDP port 8000 mentioned above, not TCP (all above mentioned ports are UDP only, not TCP) - left over from when I set up the box and "Line 2" ages ago. For RTP (http://en.wikipedia.org/wiki/Real-time_Transport_Protocol). When I was initially setting it up those were the recommended ports to forward in whatever guide I was reading. Yeah, it's probably unnecessary and I'll clean it up after it's all working.


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  #299387 16-Feb-2010 06:39
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Should all be good now, there appeared to be a issue with the the second registration for the 950 xxxx number but looks okay now.




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications

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  #299389 16-Feb-2010 06:48
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SIP ALG can cause all sorts of issues and it's sometimes better to turn it off rather than leaving it on.

You need to have the RTP ports open but don't need the whole 10000. rtp.conf contains the port range and is 10000-20000 by default.

aw



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  #299393 16-Feb-2010 07:37
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Thanks Maverick, it's all good now.

@Sbiddle: I seemed to need the ALG on when I had both my provisioned PAP2T and my Asterisk box for them both to work. Now they're both on the Asterisk box, I'll try turning it off.

Thanks again for your help guys

A

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  #299394 16-Feb-2010 07:47
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If it doesn't cause any issues then leave it. Asterisk has no NAT issues behind a firewall if you have it configured properly so there is no need for the ALG.

Bit of reading on it here


aw



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  #300487 20-Feb-2010 09:46
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Hm, I'll go back and recheck my config, but I do seem to need the ALG switched on for my cellphone's SIP client to work as a remote extension when away from home on outside WiFi networks. Without the ALG, often no audio gets to the remote extension.

Besides that, tests show everything else works regardless of the router's SIP ALG.

Anyway it's all working and I'm very happy with the setup. WAF (well, F[iancee]AF) seems good so far too...

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