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}@${domain_name}"/> </condition> I have tried many configure but not success. PLease help to fix this issue. I am veny new in this system and tried many cases to fix it. Thank all for suggesttion. ThanhTruong
Did you enable any nat helpers on your internal profile? https://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport https://wiki.freeswitch.org/wiki/NAT_Traversal Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of ThanhTruong Sent: Monday, September 16, 2013 12:48 PM To: FreeSWITCH-users@lists.freeswitch.org Subject: [Freeswitch-users] NAT issue from client Hi all, I have built basic FS server with default sip account 1011 & 1012. I have registered 2 this accounts with xlite, portgo. them in NAT via my router. When I make call, call connect and drop after 30 seconds. my dialplan is: <extension name="public_extensions"> <condition field="destination_number" expression="(\d+)$" > <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
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
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
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
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
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
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
Dear Cal Leeming, I have enabled all log level as your email. create pastebin and please have a look on it. Hope you can see some thing in this. http://pastebin.freeswitch.org/21413 thank you 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 11:27 PM Subject: Re: [Freeswitch-users] NAT issue from client 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
_________________________________________________________________________ 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
Hi, Just sharing my experience in using Freeswitch. Usually the problem of 30 s disconnect because the FS did not get the ACK after INVITE/SIP 200 OK. The FS will automatically disconnect if the timer what wait for ACK is finished (30 s). I always check for this problem if show up after new installation because it tell me that my config is not correct for that network. I have stumble to this problem many times. Usually when you register the softphone/client behind the NAT (thus using internal profile, not external profile). I fixed that by changing config in internal profile into this: <param name="rtp-ip" value="192.168.2.155"/> <param name="sip-ip" value="192.168.2.155"/> <param name="ext-rtp-ip" value="autonat:184.249.xxx.xxx"/> <param name="ext-sip-ip" value="autonat:184.249.xxx.xxx"/> the 184.249.xxx.xxx is the public static IP address (xxx to protect the innocent :) ) This configuration works for me. I hope it helps.. Donny On Sep 17, 2013 11:49 PM, "ThanhTruong" <ngoc217thanh@yahoo.com> wrote:
Dear Cal Leeming,
I have enabled all log level as your email. create pastebin and please have a look on it. Hope you can see some thing in this
http://pastebin.freeswitch.org/21413
thank you 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 11:27 PM *Subject:* Re: [Freeswitch-users] NAT issue from client
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
_________________________________________________________________________ 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
Hi all, Thank all for suggesting, But I cant fix it in internal profile. So, I have changed to register external sip client into external profile. and call works fine. I think this is my mistake in this situation. And I decide to use external profile (port 5080) for this. Thank all again for helping me. ThanhTruong ________________________________ From: Donny Hardyanto <hardyanto.donny@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, September 18, 2013 1:38 AM Subject: Re: [Freeswitch-users] NAT issue from client Hi, Just sharing my experience in using Freeswitch. Usually the problem of 30 s disconnect because the FS did not get the ACK after INVITE/SIP 200 OK. The FS will automatically disconnect if the timer what wait for ACK is finished (30 s). I always check for this problem if show up after new installation because it tell me that my config is not correct for that network. I have stumble to this problem many times. Usually when you register the softphone/client behind the NAT (thus using internal profile, not external profile). I fixed that by changing config in internal profile into this: <param name="rtp-ip" value="192.168.2.155"/> <param name="sip-ip" value="192.168.2.155"/> <param name="ext-rtp-ip" value="autonat:184.249.xxx.xxx"/> <param name="ext-sip-ip" value="autonat:184.249.xxx.xxx"/> the 184.249.xxx.xxx is the public static IP address (xxx to protect the innocent :) ) This configuration works for me. I hope it helps.. Donny On Sep 17, 2013 11:49 PM, "ThanhTruong" <ngoc217thanh@yahoo.com> wrote: Dear Cal Leeming,
I have enabled all log level as your email. create pastebin and please have a look on it. Hope you can see some thing in this
http://pastebin.freeswitch.org/21413
thank you 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 11:27 PM Subject: Re: [Freeswitch-users] NAT issue from client
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
_________________________________________________________________________ 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
participants (5)
-
Cal Leeming [Simplicity Media Ltd]
-
Donny Hardyanto
-
Moishe Grunstein
-
Paul
-
ThanhTruong