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LGE
LGE
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Hi,

I'm running an Elastix server inside my network. From between the LAN I can make calls but can't connect either to a SIP server, out of my network, or clients connect to my Elastix server.

My DSL modem makes DMZ to the ClearOS, so the it has the public IP address and all ports opened.

I currently forwarded SIP ports to the Elastix server (UDP 5060-5070), but no success.

What do you think is the problem?

LG
Friday, July 30 2010, 11:59 PM
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  • Accepted Answer

    LGE
    LGE
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    Wednesday, August 04 2010, 05:30 AM - #Permalink
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    Hi,

    Yep, I already read that page in my first attempts, both on sip_nat and sip_general_custom, just in case, but no luck :S

    I will keep reading comments on that post at freepbx. I also set up an IAX extension and forward the iax port, but still no success, I'm kind of loosing my mind as every time I check the configuration at the firewall and the Asterisk server, I do know everything should work.

    Thanks,

    LG
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  • Accepted Answer

    dpanesso
    dpanesso
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    Sunday, August 01 2010, 05:06 AM - #Permalink
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    What exactly is the problem? Can your sip trunks register?

    I have an asterisk server running inside my network and there is no need to DMZ the machine. Just port forward the correct ports. SIP on port 5060 and the RTP ports, on my config I have them 10000 to port 10020.


    You can set the RTP ports in the asterisk config file rtp.conf.


    David
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  • Accepted Answer

    LGE
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    Sunday, August 01 2010, 06:54 AM - #Permalink
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    Hi,

    Thanks for your answer. Actually the DMZ is just for the ClearOS machine and just to get into opening ports much easier.
    I can't either register to my VoIP service from between my network (or as a SIP Trunk from Elastix) or register from outside, Internet, to my Elastix server.

    I'm currently forwarding UDP ports 5060-5070 and default RTP ports (I guess 10000-20000). Are these ports of special kind of protocols? If I open those ports to the Elastix server, none would be able to use any other SIP application from between the network?

    If I connect my Mac directly to my modem and try to open my VoIP application, it succesfully registers.

    LG
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  • Accepted Answer

    dpanesso
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    Sunday, August 01 2010, 11:01 PM - #Permalink
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    I don't see why the trunks would not register. Why do you port forward ports 5060-5070?

    Me setup is to just port forward 5060 for the trunks and 10000-10200 for RTP. and of course open the ports in the incoming firewall on the ClearOS server.

    When you say you can not register from outside, Internet, to your Elastix server, do you mean a remote extension? Because if this is the case you need to config asterisk to help with the NAT. look for remote extensions on google. The config file is something like sip_nat.conf where you tell asterisk what your external IP or hostname is, what your local subnet is and some other info to help asterisk with the NAT.

    Hope that helps.

    David
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  • Accepted Answer

    LGE
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    Sunday, August 01 2010, 11:47 PM - #Permalink
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    Hi, I forwarded 5060-5070 just in case, as I could not register with my VoIP provider (Vozia.com) with just port 5060, I decided to open the whole range. Today I called my provider and it's right that they just use the port 5060, so I'm going to leave it just as 5060 in the firewall.

    I see you're opening the ports in the "Incoming" section too. I thought this section was just for ClearOS internal services (Web, FTP, SSH, etc.). Let me try just in case.

    I'll take a look in Google for the remote extension, as I guess there's something that the Asterisk server cannot "validate" when NAT is working.

    Thanks!

    LG
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  • Accepted Answer

    LGE
    LGE
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    Tuesday, August 03 2010, 09:15 PM - #Permalink
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    Hi, again

    I think it was a kind of bug in my machine, maybe corrupted files, as I updated yesterday my ClearOS and SIP successfuly registers and calls are right.

    My problem still is connecting from outside to my Elastix box. I try but never registers.

    Is any other thing I'm not setting up right?

    Thanks,

    LG
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  • Accepted Answer

    dpanesso
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    Wednesday, August 04 2010, 03:25 AM - #Permalink
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    Hello,

    Like I said in my earlier post, you need to setup some configs in the asterisk server for it to work with NAT.

    Link

    In the link above there is some good info on how to do this, it worked great for me.


    David
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  • Accepted Answer

    Sunday, January 03 2016, 04:58 PM - #Permalink
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    Ever though of VPN on from you SIP Phone also add security as certificates will be used??
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