[Freeswitch-users] NAT issue from client

[Cal Leeming [Simplicity Media Ltd]]

Many routers/phones have problems because of their loose interpretation of
SIP.

So, here's what you need to do;

* Perform a test call reproducing the problem from start to finish, collect
as many debug logs as possible, see [1], [2], [3]
* Ensure to enable SIP trace in [3]
* Attempt to find the problem yourself, if you're still having a problem
then pastebin the results [4]

I cannot stress how often this problem comes up, and you are unlikely to
get a response unless you have clearly shown to be doing your homework.

Please make sure you include (in detail) what you have attempted to do to
fix the problem, what your own thoughts are, and as much info as possible.

Hope this helps

Cal

[1] https://wiki.freeswitch.org/wiki/Debugging_Freeswitch
[2] http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch
[3] http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP
[4] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin


On Tue, Sep 17, 2013 at 5:32 AM, ThanhTruong <ngoc217thanh at yahoo.com> wrote:

> 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
>
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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