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31 posts

Geek


Topic # 196075 18-May-2016 13:50
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Hi guys

 

 

 

I am using the official FreePBX, outbound calls and internal calls work fine, but no audio for inbound calls. Actually the external public phone was still ringing after I picked up my sip phone.

 

 

 

here is my log:

 

http://pastebin.com/YmpkrstU

 

 

 

I have used a softphone ZOIPER to test my router and firewall, I can make outbound calls and received inbound calls. So I assume my router is setup correctly. The problem is in my FreePBX settings.

 

 

 

There is no port forwarding in my router as I would like to register my FreePBX as softphone to 2talk. My router is SonicWall. (I don't have an option to change my router at this stage)

 

 

 

Here is my trunk setting:

 

type=peer
username=028xxxxxxxxxx
fromuser=028xxxxxxx
secret=xxxxxxxxx
host=sip.2talk.co.nz
context=from-trunk
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw&G722
nat=no
canreinvite=no
insecure=invite,port
qualify=200

 

028xxxxxxx:xxxxxxxxx@sip.2talk.co.nz/028xxxxxxx

 

 

 

under chan sip settings -> nat settings

 

NAT = yes

 

IP config = PUBLIC IP

 

 

 

Everything else should be at defaults settings, I do changed my email setting to send some custom message, but I don't think it have anything to do with my FreePBX not getting audio from inbound calls.

 

 

 

Any thoughts?? Thanks guys

 

 

 

James

 

 

 

 


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cisconz
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  Reply # 1554816 18-May-2016 13:57
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It is a NAT issue - does your router have SIP ALG?





Hmmmm


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  Reply # 1554819 18-May-2016 14:03
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Oh no, not your SonicWall again!

 

This is certainly an issue with your firewall. Refer your reply # 1549031 at this link:

 

http://www.geekzone.co.nz/forums.asp?forumid=43&topicid=195809

 

Might pay to get your friend back again...

 

 

 

 


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  Reply # 1554866 18-May-2016 14:56
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Your issue is NAT related.

 

Since your PBX is behind a NAT firewall your trunk should be set to NAT=yes if that doesn't fix it you need to get a new firewall or a Sonicwall expert to help you. They are truly horrible firewalls.

 

 

 

 

 

 

 

 

 

 




31 posts

Geek


  Reply # 1554936 18-May-2016 15:53
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cisconz:

 

It is a NAT issue - does your router have SIP ALG?

 

 

I think this is the fix for my inbound call no audio problem. According to google, in SonicWall, SIP ALG = SIP Transformation.

 

But it seem to lead to another problem. After disabling SIP Transformation, I don't get reliable inbound calls connection. What I mean is, sometime I can't reach my sip phone. The public phone has busy/unreachable/no answer tone. My public phone can get to my sip phone most of the time tho.

 

Now changed my nat=no to nat=yes, hope it fix it. Will give you guys an update tomorrow.

 

 

 

Cheers

 

James




31 posts

Geek


  Reply # 1554939 18-May-2016 15:57
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speed:

 

Oh no, not your SonicWall again!

 

This is certainly an issue with your firewall. Refer your reply # 1549031 at this link:

 

http://www.geekzone.co.nz/forums.asp?forumid=43&topicid=195809

 

Might pay to get your friend back again...

 

 

 

 

I know, I would get my friend back to have a look. But he is on a plane to a warmer side of this planet right now... super jealous!! 




31 posts

Geek


  Reply # 1554942 18-May-2016 15:59
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sbiddle:

 

Your issue is NAT related.

 

Since your PBX is behind a NAT firewall your trunk should be set to NAT=yes if that doesn't fix it you need to get a new firewall or a Sonicwall expert to help you. They are truly horrible firewalls.

 

 

 

 

My inbound call no audio problem is fixed by disabling SIP TRANSFORMATION in SonicWall, but lead to a new problem as mentioned above. Already changed nat=yes to test, will update tomorrow.

 

 

 

Cheers


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  Reply # 1554966 18-May-2016 16:19
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jms042016:

 

But it seem to lead to another problem. After disabling SIP Transformation, I don't get reliable inbound calls connection. What I mean is, sometime I can't reach my sip phone. The public phone has busy/unreachable/no answer tone. My public phone can get to my sip phone most of the time tho.

 

 

This won't be a new problem, it will be related to the UDP timeout on the Sonicwall. You will need to understand how this is configured on the Sonicwall and whether you will need to look at ways to get around this by the use of SIP NOTIFY.

 

The ultimate solution is to get a better firewall. I wouldn't wish a Sonicwall on my worst enemy which is why I refuse to ever a deploy a VoIP setup with one.

 

 

 

 

 

 




31 posts

Geek


  Reply # 1555426 19-May-2016 11:22
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Hi 

 

 

 

I follow the instruction on this post here, it seems to fix my inbound call problem. So far I have only tested for 3 hours, so far so good. Hope this can help someone.

 

 

 

http://pbxinaflash.com/community/threads/former-sonicwall-resource-by-hbonath.12549/

 

Note:

 

I don't seem any traffic on Asterisk RTP UDP PORT 10000 - 20000, it this normal?

 

 

 

And, what other ports 2talk need? If I only open 5060 and 5061 froom LAN->WAN, I can't register with 2talk properly. I need to open all ports from LAN->WAN in order to register with 2talk successfully. Ideally, I only want to open what is needed, and no more.

 

 

 

Thanks James


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  Reply # 1555486 19-May-2016 12:18
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You really shouldn't have to port forward to get this working. It is not a very good thing to do unless you like countless hackers attempting to break in to your FreePBX server. Take a look at the asterisk logs and you will see many attempts are probably being made already.

 

SIP registration with proper NAT configuration is just fine. 

 

But to answer your question, you only need 5060 to register a device. 10000-20000 is the standard rtp range that asterisk uses, again, you shouldn't port forward these as it gives hackers yet another way to overrun your box.

 

If you must port forward at least lock it down to just the 2talk IP address (27.111.14.65 for registration).




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Geek


  Reply # 1555627 19-May-2016 14:36
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chevrolux:

 

You really shouldn't have to port forward to get this working. It is not a very good thing to do unless you like countless hackers attempting to break in to your FreePBX server. Take a look at the asterisk logs and you will see many attempts are probably being made already.

 

SIP registration with proper NAT configuration is just fine. 

 

But to answer your question, you only need 5060 to register a device. 10000-20000 is the standard rtp range that asterisk uses, again, you shouldn't port forward these as it gives hackers yet another way to overrun your box.

 

If you must port forward at least lock it down to just the 2talk IP address (27.111.14.65 for registration).

 

 

 

 

Hi chevrolus,

 

I think I use the wrong words. I didn't not forward any ports to the great wild word. What I really mean is I set my firewall to allow my PBX (behind NAT) to talk to 2talk through 5060 - 5061, and 10000 - 20000, but not the other way around. And in my firewall rules, it only allow those ports to 2talk IP address. (Well at least that's what I intent to)

 

But just in case I made a mistake (PBX and networking is new to me), where is the asterisk logs file I should take a look at?

 

Thanks

 

 

 

James

 

 

 

 


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  Reply # 1555660 19-May-2016 15:19
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Ahh sorry. Understood. In the 'reports' section, there is 'asterisk logfiles'.

 

 


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