- FreeSWITCH-users - FreeSWITCH Mailing Lists

yahoo blocking.
by Michael Jerris 31 Mar '08

31 Mar '08

Notice to yahoo users. We have been having ongoing issues with the mailing lists with yahoo, even after going through their white-list procedures. We suggest moving your mailing list subscriptions to a different e-mail account as yahoo seems to be rejecting or delaying many list emails. Also, please do not report list messages as spam. If you have any questions, feel free to reply privately to me. Thanks Mike

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 21, Issue 76
by Ritesh Singh 31 Mar '08

31 Mar '08

Thanks to Brian and Michael for there valuable help. While trying with the "echo" from default.xml, I was getting a beep like sound, so i assume that my port audio is working.( I didn't hear the word echo though..is it fine ?). Then i tried to play a mp3 file and then a wav file BY adding few changes. <extension name="echo"> <condition field="destination_number" expression="^9996$"> <action application="answer"/> <action application="send_display" data="Echo Test"/> <action application="playback" data="file location"/> <action application="echo"/> </condition> </extension> When i tried this, freeswitch was not able to play the file as it was not able to find the ext. So what i am supposed to do so that i am able to play wav or mp3 file from specified location. I have one more question that can get me kick start in decdoing the freeswitch code. Here it goes: 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y.(Currently assumng that both the ips are on same lan), Then what changes i am supposed to do in default.xml so that i can bridge the call between x.x.x.xand y.y.y.y using the freeswitch both computer can talk . I am using mod_portaudio as my end point. I think, i need to have freeswitch running at both the computer as how can y.y.y.y know that x.x.x.x has called it. Can you please tell me the exact dialplan to achieve this. And also is it possible to this with mod portaudio or i am in totally worng direction. Please suggest. (My irc name is Ritesh. I have just joined the IRC) Regards Ritesh On Sat, Mar 29, 2008 at 12:24 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: New to FreeSwitch (Ritesh Singh) (Ritesh Singh) > 2. Re: New to FreeSwitch (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 28 Mar 2008 22:18:47 +0530 > From: "Ritesh Singh" <riteshsingh81(a)gmail.com> > Subject: Re: [Freeswitch-users] New to FreeSwitch (Ritesh Singh) > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <98a09a460803280948q56f005f6w129adcc4fb6f941a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Fri, Mar 28, 2008 at 9:30 PM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > > > Send Freeswitch-users mailing list submissions to > > freeswitch-users(a)lists.freeswitch.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > or, via email, send a message with subject or body 'help' to > > freeswitch-users-request(a)lists.freeswitch.org > > > > You can reach the person managing the list at > > freeswitch-users-owner(a)lists.freeswitch.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of Freeswitch-users digest..." > > > > > > Today's Topics: > > > > 1. Re: New to FreeSwitch (Brian West) > > 2. Re: How to bridge 2 sessions with Javascript? (Dale Thatcher) > > 3. Re: How to bridge 2 sessions with Javascript? (Anthony Minessale) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Fri, 28 Mar 2008 09:57:23 -0500 > > From: Brian West <brian.west(a)mac.com> > > Subject: Re: [Freeswitch-users] New to FreeSwitch > > To: freeswitch-users(a)lists.freeswitch.org > > Message-ID: <CE2A83CB-29B3-41B9-BD01-DFD7861AC801(a)mac.com> > > Content-Type: text/plain; charset="us-ascii" > > > > > > On Mar 28, 2008, at 8:07 AM, Ritesh Singh wrote: > > > > > Hi All, > > > > > > I am very new to freeswitch. It will be great if some one can tell > > > me few things: > > > > Welcome to the community. > > > > > > > > > > > 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. > > > Then what changes i am supposed to do and at what place. I would > > > like to use the mod_portaudio for this purpose. > > > > > > 2) I started the windows freeswitch and loaded port_audio by using > > > "load mod_portaudio" , then used "pa call 1234"...the session gets > > > initiated but the call gets hanged up with the log "portaudio/1234 > > > [CS_RING] [NO_ROUTE_DESTINATION]". > > > > I'm sure this message is rather clear... You do not have a route in > > your dialplan for 1234 thus it fails. > > > > > 3) Suppose i have a jabber server x.net. and i have 2 users having > > > account at that jabber server. Then how can i use mod_dingaling so > > > that i have the voice chat between those user of the jabber server > > > x.net. > > > Also, i would like to dump the all the record regarding the call > > > made, like duration of call , person who initiated the call, call > > > destination , etc. > > > > Feels like your trying to do too much at once before you fully > > understand what is going on. > > > > > Any help in this regard is highly appreciable. Please do forgive > > > me if you think its stupid mail but i am desparate for these answers. > > > > > > > You're trying to fly before you can even crawl. You have many things > > to learn. > > > > Best things to do are start here http://wiki.freeswitch.org and > > #freeswitch on irc.freenode.net > > > > Remember I'll be very glad to answer questions but the requirement is > > that you MUST put the info you learn on the wiki and pay it forward to > > others. > > > > Thanks, > > /b > > > > > > > >

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Help Setting up Freeswitch
by saumya garg 31 Mar '08

31 Mar '08

The following is my scenario: I have some SIP account, xyzUSER(a)ekiga.net, registered on Ekiga. I have a 10 digit US number that is attached to this SIP account. I want Freeswitch to answer incoming calls on this account. I am able to use the ekiga client that comes with ubuntu and connect to the server and receive calls using this 10 digit number. The following is included in my /conf/sip_profiles/outbound folder: <include> <gateway name="ekiga.net"> <param name="username" value="xyzUSER"/> <param name="from-user" value="xyzUSER"/> <param name="password" value="xyzPASS"/> <param name="extension" value="xyzUSER"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> I've tried the above settings with and without the settings "register" and "caller-id-in-from" commented. When freeswitch starts, I know that these settings are at least partially correct, for I see the following in my log file: 2008-03-29 20:17:41 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ekiga.net 2008-03-29 20:17:41 [DEBUG] sofia_reg.c:753 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"ekiga.net":xyzUSER:xyzPASS'. 2008-03-29 20:17:41 [DEBUG] sofia_reg.c:684 sofia_reg_handle_sip_r_register() Changing expire time to 3600 by request of proxy sip:ekiga.net 2008-03-29 20:17:42 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 20:37:12 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 21:15:35 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 21:59:21 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net 2008-03-29 22:42:26 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered ekiga.net In spite of these log messages, when I try to dial my 10 digit number I only hear a busy signal. I feel that the error is in dialplan. As per the diagram located at http://wiki.freeswitch.org/wiki/Image:FreeswitchDiagram.png, which I created following a link that was provided to me on freenode via IRCt, calls coming in via my ekiga SIP should end up in the secure default context. Nevertheless, I have adding the following block of code to both my public and default contexts: <extension name="xyzUSER"> <condition field="destination_number" expression="^xyzUSER$"> <action application="answer"/> <action application="echo"/> <anti-action application="answer"/> </condition> </extension> What else could I be missing? Thanks in advance for your help! Sincerely, Saumya Garg

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Sip Gateway registration - User Agent Change
by Joseph Bajin 31 Mar '08

31 Mar '08

In order to register with my sip provider, I need to have a specific user agent. Can someone tell me how I change that in my setup? Thank You Joe

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Segmentation fault on mod_python, with trunk r7982
by Arnaldo de Moraes Pereira 30 Mar '08

30 Mar '08

Hello, I've found a segfault running python scripts through mod_python and filled a bug on jira: http://jira.freeswitch.org/browse/MODLANG-54 Take care, Arnaldo M Pereira

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Sun T2000 Fund Raiser...
by Brian West 29 Mar '08

29 Mar '08

Ok guys we are trying to raise some funds to buy an 8 core T2000 box for the FreeSWITCH project. Anyone wishing to assist in this endeavor please email me off list. Thanks, Brian West brian(a)freeswitch.org

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Basic information and help needed
by Abilash Praveen 29 Mar '08

29 Mar '08

Hello, I'm new to freeswitch and VoIP on the whole. I'm trying to deploy Freeswitch for a LAN. My current senerio is this: I have a CentOS linux server set up and have about 5 PCs on the LAN. The Linux box is the gateway to the PCs. I want to set up Freeswitch and have phones on all the computers. I want to use Freeswitch as a softPBAX to route calls to all the five persons and also to be able to call the five persons internally. Eg, if someone calls our office number, it should be picked up by person 1. He will then forward the calls to any of the 5 persons in the LAN. Where do I start? I have moderate knowledge in Linux and PHP so should be able to work with Freeswitch. But my question is how can i input the telephone line (from my provider) into the server? Is it also possible to use ordinary phones across LAN instead of SIP phones (I assume it is software phones that will use the server's IP to get hooked in). I'm sorry if my questions sound too foolish, but since I'm totally new to VoIP I'm having problems to kick start. Thank you in advance for your help. Kind regards, Abilash Praveen

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FreeSwitch on FreeBSD
by Mark Crane 28 Mar '08

28 Mar '08

I'm trying to get FreeSwitch installed on FreeBSD 7. Right now trying to find all the dependencies. * SVN: Lots of choices here. * GNUMAKE: The GNU version of make. gmake-3.81_2 (required by autoconf-2.61_2) * AUTOCONF: Version 2.50 or higher ports/devel/autoconf261/ autoconf-2.61_2 * GCC: Version 3.95 or higher ports/lang/gcc34/ gcc-3.4.6_3,1 * WGET: Almost any version pkg_add -r wget * AUTOMAKE: Version 1.70 or higher unknown * GLIBTOOL: Version 1.5.14 or higher unknown (maybe the same as libtool 1.5.24?) pkg_add -r libtool * LIBNCURSES: Almost any version unknown (is this the same as ncurses-5.6_1?) pkg_add -r ncurses So right now I'm needing to find automake, and confirm if gmake, libtool, and ncurses are correct. Any help is appreciated once I have something working I will work on getting it added to the wiki. In the long run I'm planning on creating a package for freeswitch to run on a PFSense Appliance as add on. (PFSense Appliance is a new project coming out of PFSense firewall project except the appliance will be for more general purpose use). Best Regards, Mark J Crane ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-users] New to FreeSwitch (Ritesh Singh)
by Ritesh Singh 28 Mar '08

28 Mar '08

On Fri, Mar 28, 2008 at 9:30 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: New to FreeSwitch (Brian West) > 2. Re: How to bridge 2 sessions with Javascript? (Dale Thatcher) > 3. Re: How to bridge 2 sessions with Javascript? (Anthony Minessale) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 28 Mar 2008 09:57:23 -0500 > From: Brian West <brian.west(a)mac.com> > Subject: Re: [Freeswitch-users] New to FreeSwitch > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <CE2A83CB-29B3-41B9-BD01-DFD7861AC801(a)mac.com> > Content-Type: text/plain; charset="us-ascii" > > > On Mar 28, 2008, at 8:07 AM, Ritesh Singh wrote: > > > Hi All, > > > > I am very new to freeswitch. It will be great if some one can tell > > me few things: > > Welcome to the community. > > > > > > > 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. > > Then what changes i am supposed to do and at what place. I would > > like to use the mod_portaudio for this purpose. > > > > 2) I started the windows freeswitch and loaded port_audio by using > > "load mod_portaudio" , then used "pa call 1234"...the session gets > > initiated but the call gets hanged up with the log "portaudio/1234 > > [CS_RING] [NO_ROUTE_DESTINATION]". > > I'm sure this message is rather clear... You do not have a route in > your dialplan for 1234 thus it fails. > > > 3) Suppose i have a jabber server x.net. and i have 2 users having > > account at that jabber server. Then how can i use mod_dingaling so > > that i have the voice chat between those user of the jabber server > > x.net. > > Also, i would like to dump the all the record regarding the call > > made, like duration of call , person who initiated the call, call > > destination , etc. > > Feels like your trying to do too much at once before you fully > understand what is going on. > > > Any help in this regard is highly appreciable. Please do forgive > > me if you think its stupid mail but i am desparate for these answers. > > > > You're trying to fly before you can even crawl. You have many things > to learn. > > Best things to do are start here http://wiki.freeswitch.org and > #freeswitch on irc.freenode.net > > Remember I'll be very glad to answer questions but the requirement is > that you MUST put the info you learn on the wiki and pay it forward to > others. > > Thanks, > /b > > > >

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Re: [Freeswitch-users] How to bridge 2 sessions with Javascript?
by Anthony Minessale 28 Mar '08

28 Mar '08

and just to point it out, you can also do: session1 = new Session(); session1.originate(session1, "{ignore_early_media=true}sofia/gateway/ asterlink.com/19184249378"); session1.execute("bridge", "sofia/gateway/asterlink.com/19184238080"); or even better // this will transfer the channel into the dialplan and end the script // thus reducing overhead of leaving JS open. session1.execute("transfer", "19184238080"); Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Brian West <brian.west(a)mac.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 27, 2008 9:55:25 PM Subject: Re: [Freeswitch-users] How to bridge 2 sessions with Javascript? Revised script: session1 = new Session(); session1.originate(session1, "{ignore_early_media=true}sofia/gateway/ asterlink.com/19184249378"); session2 = new Session(); session2.originate(session2, "sofia/gateway/asterlink.com/19184238080"); bridge(session1, session2); session.hangup(); while (session1.ready() && session2.ready()) { } You don't need waitForAnswer because the ignore_early_media=true performs that automatically for you so it won't return till you answer. And you don't want the ignore_early_media on the second leg otherwise you get NO ringback and its just silent till the other end is answered. /b On Mar 27, 2008, at 9:10 PM, Nicolas Brenner wrote: > Hello everybody again, > > First of all I want to say thanks to the people on this list and on > IRC, I'm really surprised (in a very good way) of the help I've > received. > > Now to my problem: I'm trying to bridge two SIP calls together with a > JS script, and to achieve it, I'm did the following: > - created a dialplan entry for extension 500 which calls js script > - created a js script with the following code: > > // Create new_session > session1 = new Session(); > session1.originate(session, > "{ignore_early_media=true}sofia/gateway/sip.sipdiscount.com/ > 005624949458"); > session1.waitForAnswer(10000); > > new_session = new Session(); > new_session.originate(session, > "{ignore_early_media=true}sofia/gateway/sip.sipdiscount.com/ > 0056979039388", > 30); > new_session.waitForAnswer(10000); > > // IF everybody is ready, then bridge our current session & the > new_session > if (session1.ready() && new_session.ready()) { > console_log("info", "Interoligofrenico!\n"); > bridge(session1, new_session); > } > > // hangup when done > session1.hangup(); > new_session.hangup(); > > When I register with extension 1000 using a softphone and dial > extension 500, the code above successfully creates the two new > sessions, makes the calls in order, and supposedly bridges the > sessions (I get no error about the bridge on the console, and I get > the log text too), but I get no audio on either end. Anybody know > what's wrong with the code? > > Btw, how can I originate a call to a configure extension? All examples > on the wiki use the 'sofia syntax', should I just use > sofia/default/1001 for example? > > Thanks! > > Nicolas > > _______________________________________________ > 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 _______________________________________________ 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 ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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