[Freeswitch-users] NAT issue from client

[ThanhTruong]

HI all,

In my case, my FS server is in public IP. and I see that I am getting nat in client (sip phone) side.
I have tried with sturn server, but no luck, I have checked contact detail in database and see that it can get external IP correct.

I have read much, tried more than 2 weeks with this, but cant get it works :( 

That is a bad situation for me.
:( 
ThanhTruong


________________________________
 From: Paul <pasha at prosperity4ever.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, September 17, 2013 12:40 AM
Subject: Re: [Freeswitch-users] NAT issue from client
 


Edit your public sip profile xml and make sure that your internal ip of the server and external ip of the server (for both rtp and sip) are set properly. That should fix it.

Paul

On Mon, 16 Sep, 2013 at 10:02 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:

Hello,
>
>
>You'll see this has been raised many times already, a quick google search shows;
>
>
>https://www.google.co.uk/search?q=freeswitch+30+seconds
>
>http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073842.html
>
>http://jira.freeswitch.org/browse/FS-3496
>
>http://freeswitch-users.2379917.n2.nabble.com/NAT-issues-Outbound-Call-drops-after-30-seconds-td7386901.html
>
>http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-seconds-issue-FreeSWITCH-users-Digest-Vol-77-Issue-117-td7584663.html
>
>
>
>Please review all those answers and let us know if it resolves your problem.
>
>
>If not then please update the ticket with some debug logs (remember to paste into pastebin, not direct into the ticket)
>https://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch
>
>
>Hope this helps
>
>
>Cal
>
>
>
>On Mon, Sep 16, 2013 at 5:48 PM, ThanhTruong <ngoc217thanh at yahoo.com> wrote:
>
>Hi all,
>>
>>
>>I have built basic FS server with default sip account 1011 & 1012.
>>I have registered 2 this accounts with xlite, portgo. them in NAT via my router.
>>When I make call, call connect and drop after 30 seconds.
>>
>>
>>my dialplan is:
>>
>>
>><extension name="public_extensions">
>>      <condition field="destination_number" expression="(\d+)$" >
>>        <action application="set" data="hangup_after_bridge=true"/>
>>        <action application="bridge" data="sofia/internal/${destination_number}@${domain_name}"/>
>>      </condition>
>>
>>
>>
>>
>>I have tried many configure but not success.
>>PLease help to fix this issue.
>>
>>
>>I am veny new in this system and tried many cases to fix it.
>>
>>
>>Thank all for suggesttion. 
>>ThanhTruong
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




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