[Freeswitch-users] NAT issue from client

Lloyd Aloysius lloyd.aloysius at gmail.com
Tue Sep 17 20:45:11 MSD 2013 

Set the following in the user vars for the 1001 and reload and test
https://wiki.freeswitch.org/wiki/NAT_Traversal

sip_force_contact =NDLB-connectile-dysfunction

NDLB-received-in-nat-reg-contact = true


*
*


On Tue, Sep 17, 2013 at 12:29 PM, ThanhTruong <ngoc217thanh at yahoo.com>wrote:

> This is log in FS_CLI when i get siptrace and log level.
> http://pastebin.freeswitch.org/21412
>
> my router is Smartel network, this is as another router and open for all.
> I have tried with many sip server as asterisk, and it works fine.
>
> :(
>
> Thanks,
>
> ThanhTruong
>
>   ------------------------------
>  *From:* Lloyd Aloysius <lloyd.aloysius at gmail.com>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, September 17, 2013 10:55 PM
>
> *Subject:* Re: [Freeswitch-users] NAT issue from client
>
> sofia global siptrace on and capture the log and post in paste bin.
>
> What is the made/model of the router?
> *
> *
>
>
> On Tue, Sep 17, 2013 at 11:22 AM, ThanhTruong <ngoc217thanh at yahoo.com>wrote:
>
> Hi All,
>
> I have tried to build new server with GIT version, and current version of
> FS.
>
> Then, I have disable SIP ALG in my router too.
>
> I have make a full test call from 1011 to 1012 to check with this issue,
> used ngrep to capture all package send and receive.
>
> http://pastebin.freeswitch.org/21411
>
> my FS ip is 123.123.123.123.
>
> The call drop after 30 seconds(to 32 seconds).
>
> I see that when FS get 200OK, it send ACK to 1011, but it keep get 200OK
> from client.
>
> Hope some one can help me on this case.
>
> my dialplan :
>
> <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>
>     </extension>
>
> Thank all in advance.
>
> ThanhTruong
>
>   ------------------------------
>  *From:* Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, September 17, 2013 5:14 PM
>
> *Subject:* Re: [Freeswitch-users] NAT issue from client
>
> 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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>
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>
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>
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>
> 
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