Thanh, Can you please use the debugging/troubleshooting steps mentioned in my previous email, then pastebin the console log output. Specifically; # sofia global siptrace on # sofia loglevel all 9 # sofia tracelevel alert # console loglevel debug # fsctl debug_level 10 Cal On Tue, Sep 17, 2013 at 4:22 PM, ThanhTruong <ngoc217thanh@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@simplicitymedialtd.co.uk>
*To:* FreeSWITCH Users Help <freeswitch-users@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@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@prosperity4ever.com> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Cc:* FreeSWITCH Users Help <freeswitch-users@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@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...
http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-s...
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@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org