- FreeSWITCH-users - FreeSWITCH Mailing Lists

how to encrypt the sip message?
by 彭勇 17 Sep '13

17 Sep '13

one of my sip gateway using a encrypted SIP message. char * sip_encrypt(char* str) { ... return encrypted_str; } i should send encrypted to the gateway. the gateway also send me a encrypted SIP message, and i should decrypt it. then how can i do it in Freeswitch? where can i code insert sip_encrypt()? thanks -- Peng Yong

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Re: [Freeswitch-users] NAT issue from client
by Lloyd Aloysius 17 Sep '13

17 Sep '13

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(a)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(a)gmail.com> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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(a)simplicitymedialtd.co.uk> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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(a)prosperity4ever.com> > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Cc:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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-drop… > > http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-… > > 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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] NAT issue from client
by Lloyd Aloysius 17 Sep '13

17 Sep '13

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(a)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(a)simplicitymedialtd.co.uk> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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(a)prosperity4ever.com> > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Cc:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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-drop… > > http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-… > > 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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Record file not save when call is disconnect in session:recordFile
by Sanath Prasanna 17 Sep '13

17 Sep '13

Hi, I used LUA for voicemail (vm) system. I want to save message & send message to end user about vm notification, even user disconnect call by himself when recording message. Currently if user disconnect within recording, file is not save & notification is not going. Pls help to solve that. AheS. Robert code: session:recordFile(vm_file_namCurrentle, vm_record_duration, vm_record_duration, 0); -- pressing # ends the recording

2 2

Index files failing to download while updating package repositories on my raspberry pi running Debian Wheezy
by Pranav Lal 17 Sep '13

17 Sep '13

Hi all, I am trying to download freeswitch on to my raspberry pi running debian Wheezy.. I am following the instructions for debian at http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Upgrading_an d_Re-installation Add the FreeSWITCH APT repository: (for Debian Wheezy) echo 'deb http://files.freeswitch.org/repo/deb/debian/ wheezy main' >> /etc/apt/sources.list.d/freeswitch.list PL] ok. Import the repo signing key curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - PL] This process was flawless. However, when I did a apt-get update, I got the following error. Hit http://archive.raspberrypi.org wheezy Release.gpg Hit http://raspberrypi.collabora.com wheezy Release.gpg Get:1 http://files.freeswitch.org wheezy Release.gpg [198 B] Hit http://raspberrypi.collabora.com wheezy Release Hit http://archive.raspberrypi.org wheezy Release Get:2 http://mirrordirector.raspbian.org wheezy Release.gpg [490 B] Get:3 http://files.freeswitch.org wheezy Release [3,221 B] Get:4 http://mirrordirector.raspbian.org wheezy Release [14.4 kB] Hit http://raspberrypi.collabora.com wheezy/rpi armhf Packages Hit http://archive.raspberrypi.org wheezy/main armhf Packages Get:5 http://mirrordirector.raspbian.org wheezy/main armhf Packages [7,414 kB] Ign http://raspberrypi.collabora.com wheezy/rpi Translation-en_GB Ign http://archive.raspberrypi.org wheezy/main Translation-en_GB Ign http://raspberrypi.collabora.com wheezy/rpi Translation-en Ign http://archive.raspberrypi.org wheezy/main Translation-en Hit http://mirrordirector.raspbian.org wheezy/contrib armhf Packages Hit http://mirrordirector.raspbian.org wheezy/non-free armhf Packages Hit http://mirrordirector.raspbian.org wheezy/rpi armhf Packages Ign http://mirrordirector.raspbian.org wheezy/contrib Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/contrib Translation-en Ign http://mirrordirector.raspbian.org wheezy/main Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/main Translation-en Ign http://mirrordirector.raspbian.org wheezy/non-free Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/non-free Translation-en Ign http://mirrordirector.raspbian.org wheezy/rpi Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/rpi Translation-en Fetched 7,432 kB in 16min 49s (7,358 B/s) W: Failed to fetch http://files.freeswitch.org/repo/deb/debian/dists/wheezy/Release Unabl e to find expected entry 'main/binary-armhf/Packages' in Release file (Wrong sources.list entry or malformed file) E: Some index files failed to download. They have been ignored, or old ones used instead. root@raspberrypi:/usr/local# Should I continue with the installation or should I stick to building from source? I plan to use freeswitch as a PBX for my house where I will connect a number of mobile phones using SIP. Pranav

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Re: [Freeswitch-users] NAT issue from client
by Lloyd Aloysius 17 Sep '13

17 Sep '13

Thanh, What is the made and model of your router. My recommendation is capture the trace and find the reason for the problem. Your problem is related to the network and not freeswitch. If the router have SIP ALG. Disable SIP ALG and try. Lloyd On Tue, Sep 17, 2013 at 6:14 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming(a)simplicitymedialtd.co.uk> wrote: > 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(a)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(a)prosperity4ever.com> >> *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> *Cc:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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-drop… >> >> http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-… >> >> 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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] OpenSIPs + Freeswitch + Registrations
by Chris B. Ware 17 Sep '13

17 Sep '13

Steven, you pointed me on the right direction to fix it: since I use xml_curl, there was a not closed tag and dsn was not present on internal profile. Thank you very much.

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TLS + NAT
by Vincent Xia 17 Sep '13

17 Sep '13

hi guys, both TLS and NAT are working fine with my FS system, but put them together raises problem, e.g. i can have encrypted voice calls(TLS+SRTP) within the same LAN, and uncrypted voice calls over different LANs, but when trying TLS+SRTP over different LANs, i cannot get my softphone(here im using Linphone) register to FS, it reports timeout. i have a dedicated profile for NAT as below, could anyone figure out what the problem is? <profile name="tnat"> <settings> <param name="debug" value="0" /> <param name="sip-trace" value="no" /> <param name="rfc2833-pt" value="101" /> <param name="sip-port" value="5063" /> <param name="dialplan" value="XML" /> <param name="context" value="public" /> <param name="dtmf-duration" value="100" /> <param name="codec-prefs" value="$${outbound_codec_prefs}" /> <param name="use-rtp-timer" value="true" /> <param name="hold-music" value="$${hold_music}" /> <param name="rtp-timer-name" value="soft" /> <param name="manage-presence" value="false" /> <param name="aggressive-nat-detection" value="true" /> <param name="apply-nat-acl" value="rfc1918" /> <param name="inbound-codec-negotiation" value="generous" /> <param name="nonce-ttl" value="60" /> <param name="auth-calls" value="true" /> <param name="rtp-timeout-sec" value="1800" /> <param name="rtp-ip" value="$${local_ip_v4}" /> <param name="sip-ip" value="$${local_ip_v4}" /> <param name="ext-rtp-ip" value="111.111.111.111" /> <param name="ext-sip-ip" value="111.111.111.111" /> <param name="force-register-domain" value="111.111.111.111" /> <param name="stun-enabled" value="false" /> <param name="rtp-timeout-sec" value="300" /> <param name="rtp-hold-timeout-sec" value="1800" />  <param name="tls" value="true"/>  <param name="tls-only" value="false"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="5064"/>    <param name="tls-passphrase" value=""/>  <param name="tls-verify-date" value="true"/>   <param name="tls-verify-policy" value="none"/>  <param name="tls-verify-depth" value="2"/>  <param name="tls-verify-in-subjects" value=""/>  <param name="tls-version" value="$${sip_tls_version}"/>  <param name="inbound-late-negotiation" value="true"/>  <param name="inbound-zrtp-passthru" value="true"/> </settings> </profile>

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presence
by fs 17 Sep '13

17 Sep '13

I am having trouble with blf's on the freeswitch, when a device changes its port, either due to the router or the sip device randomly changing its port, the freeswitch keeps sending notify messages to the previous port. The remainder of the signaling is ok, the devices work correctly, just the blf's do not work. I use freeswitch in a hosted mode with the ip sets behind a router/firewall. I feel I have missed a setting someplace, any assistance would be appreciated. Darcy Primrose

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Re: [Freeswitch-users] How to make callcenter not auto-answer
by Jimmy Chang 17 Sep '13

17 Sep '13

In other words, is it possbile to configure callcenter application pre_answer calls? 於 2013/9/17 上午 11:38, Jimmy Chang 提到: > Hi, > > We have configured a IVVR. > Here is the call flow: > 1) user makes a call to FS > 2) FS bridges the call to IVVR > 3) IVVR plays welcome video and brdiges the call back to FS callcenter > > My problem is when FS callcenter recieves the call, it would answer > the call and

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