March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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How to bridge 2 sessions with Javascript?
by Nicolas Brenner 28 Mar '08

28 Mar '08

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

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 21, Issue 73
by Henk Oegema 28 Mar '08

28 Mar '08

> Message: 2 > Date: Thu, 27 Mar 2008 17:05:43 -0500 > From: Brian West <brian.west(a)mac.com> > Subject: Re: [Freeswitch-users] How to reply ? > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <CF21EBC2-F296-468E-BC9B-9CE1C3ED3B3F(a)mac.com> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Press the reply button in your email program and they'll thread > automatically. > > /b > But I get all messages in a bunch, one or twice a day. So I can't reply to a single message. May be I need to change my settings? So let me see what happens to this message. (I deleted the rest of this Vol21, Issue 73)

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Getting Started Answering Calls
by Brian Tanner 28 Mar '08

28 Mar '08

Hi there. I might end up being active in this list, so I'll introduce myself. My name is Brian, I'm a complete newbie to telephony systems in general. I'm sorry, I know that can be frustrating, I don't really know the jargon and I don't really understand how things work. The good news is that I am good at figuring things out and I like to fiddle. Let me see if I have my jargon right: I have a DID mapped to AIX with a service provider (DID WorldWide). I have a cable modem at home, and my laptop (where FreeSwitch is) is the DMZ. I haven't touched the FreeSwitch config. When I call my local DID, FreeSwitch answers: freeswitch(a)bt-macBook.local> 2008-03-27 16:53:45 [NOTICE] mod_iax.c: 1011 mod_iax_runtime() Incoming call connected 17807010953, 7806692738, (null) 256/2097023 2008-03-27 16:53:45 [NOTICE] mod_iax.c:1019 mod_iax_runtime() New Inbound Channel (null)! 2008-03-27 16:53:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel IAX/17807010953-f30d [14617d63-9bb8-428b-8207-22a90693ae42] 2008-03-27 16:53:45 [INFO] mod_iax.c:399 iax_set_codec() Activate Codec PCMU/8000 20 ms 2008-03-27 16:53:45 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing ->17807010953@default 2008-03-27 16:53:45 [NOTICE] switch_ivr.c:1001 switch_ivr_session_transfer() Transfer IAX/17807010953-f30d to enum[17807010953@default] 2008-03-27 16:53:46 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting 2008-03-27 16:53:46 [NOTICE] switch_core_state_machine.c:114 switch_core_standard_on_ring() Hangup IAX/17807010953-f30d [CS_RING] [NO_ROUTE_DESTINATION] 2008-03-27 16:53:46 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 1 (IAX/17807010953-f30d) Ended 2008-03-27 16:53:46 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel IAX/17807010953-f30d [CS_HANGUP] Bus error (core dumped) So, the crash sucks, and doesn't happen every single time. I'm not tooo worried about that, because things are broken above. It looks like the call was answered (and it even knows who called it -- cool). But, it looks like FreeSwitch doesn't know what to do with the call, and so it aborts and hangs up. I think this is expected behavior. I want to use FreeSwitch for a few things, but the first is to build an IVR and Voicemail system. Can someone point me to resources or give me some quick tips on how to start this? There appear to be almost hundreds of config files, and it's all new to me, so I don't really know where to start, which files do what, etc, etc. If someone could help me with minimal instructions on how to play a sound file based on some control-flow code I write, or some way to run a script, or whatever, I'd be very grateful. I'm just really overwhelmed and I don't know how to start. Thanks in advance, very much! PS: If people help me get rolling, I'm happy to update wikis and/or write little tutorials as I go to save this hassle for future... -- Brian Tanner Ph.D Student, University of Alberta brian(a)tannerpages.com

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How to reply ?
by Henk Oegema 27 Mar '08

27 Mar '08

How do I reply to a message, so that my answer to a question remains in the same thread. (i.s.o. opening a new thread) ?

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SIP dialout problems
by Nicolas Brenner 27 Mar '08

27 Mar '08

Hi, I'm having trouble dialing out some phone numbers through a SIP gateway. I configured the gateway on conf/sip_profiles/outbound/gizmo.xml as: <include> <gateway name="gizmo"> <param name="debug" value="1"/> <param name="username" value="1747xxxxxx"/> <param name="from-domain" value="proxy01.sipphone.com"/> <param name="password" value="MYPASSWORD"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="proxy01.sipphone.com"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="retry_seconds" value="3600"/> </gateway> </include> Then I created an extension on conf/dialplan/public.xml like this: <extension name="chile"> <condition field="destination_number" expression="^0056(.*)"> <action application="bridge" data="sofia/gateway/gizmo/0056$1"/> </condition> </extension> The above catches all calls that begin with the 0056 prefix (calls to Chile). So what I do is, register with a softphone using extension 1000, then I dial 0056xxxxx, and I get a recording saying the extension I'm calling is unavailable. The weird thing is that if I don't use 0056, the call is made, but then I get a recording from Gizmo (my test gateway) saying the number is wrong (of course). If instead of an expression, I dial the fixed number 1-800-466-4411 (Google 411), the called is successfully bridged. Also if I dial a Gizmo user number (1747xxxxxx) the call works. What's the problem with 0056? Apparently the call is never "leaving" freeswitch. For an unsuccessful call, I get this on the console: 2008-03-27 12:43:59 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000@medularis01.dyndns.org:5060 [dbbaff68-0791-4253-a07c-ab2c4130df18] 2008-03-27 12:43:59 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Nico->005624949458@default 2008-03-27 12:43:59 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/outbound/005624949458 [c0b1791c-eb96-464a-bc30-5d6481ed0c10] 2008-03-27 12:44:02 [NOTICE] sofia.c:1950 sofia_handle_sip_i_state() Hangup sofia/outbound/005624949458 [CS_HOLD] [NORMAL_UNSPECIFIED] 2008-03-27 12:44:02 [INFO] mod_dptools.c:1514 audio_bridge_function() Originate Failed. Cause: NORMAL_UNSPECIFIED 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 16 (sofia/outbound/005624949458) Ended 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/005624949458 [CS_HANGUP] 2008-03-27 12:44:02 [NOTICE] mod_dptools.c:1541 audio_bridge_function() Hangup sofia/default/1000@mydomain:5060 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 15 (sofia/default/1000@mydomain:5060) Ended 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1000@mydomain:5060 [CS_HANGUP] What does NORMAL_UNSPECIFIED mean?! I found this page on the wiki: http://wiki.freeswitch.org/wiki/Hangup_causes but it's only a list of the codes. The page says "The codes are documented in src\switch_channel.c" but there's no documentation there at all, just the code. Any help would be greatly appreciated. Thanks in advance for your time and help. Regards, -- Nicolás Brenner

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