[Freeswitch-users] New to FreeSwitch (Ritesh Singh)
Ritesh Singh
riteshsingh81 at gmail.com
Fri Mar 28 09:48:47 PDT 2008
On Fri, Mar 28, 2008 at 9:30 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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> 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 at mac.com>
> Subject: Re: [Freeswitch-users] New to FreeSwitch
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <CE2A83CB-29B3-41B9-BD01-DFD7861AC801 at 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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> ------------------------------
>
> Message: 2
> Date: Fri, 28 Mar 2008 15:12:09 +0000
> From: Dale Thatcher <freeswitch at dalethatcher.com>
> Subject: Re: [Freeswitch-users] How to bridge 2 sessions with
> Javascript?
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <1206717129.8070.51.camel at desktop>
> Content-Type: text/plain
>
> Great, guess I'll drop that from my config then. Sorry Nicolas must be
> something else.
>
> - Dale
>
> On Fri, 2008-03-28 at 09:53 -0500, Brian West wrote:
> > Dale,
> > This was due to a rogue SRV record saying use TCP that has since
> been
> > corrected after a month of talking to them about it.
> >
> > /b
> >
> > On Mar 28, 2008, at 8:50 AM, Dale Thatcher wrote:
> >
> > > I had some slow connection problems with sipphone that were solved by:
> > >
> > > <param name="bind-params" value="transport=udp"/>
> > >
> > > in the sip profile, might be worth a go. BTW this is a bug with
> > > sipphone, not with Freeswitch.
> > >
> > > - Dale
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> ------------------------------------------------
> http://myhelpa.com - Sign up to be a Beta Helpa.
>
>
>
> ------------------------------
>
> Message: 3
> Date: Fri, 28 Mar 2008 08:16:50 -0700 (PDT)
> From: Anthony Minessale <anthmct at yahoo.com>
> Subject: Re: [Freeswitch-users] How to bridge 2 sessions with
> Javascript?
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <539060.7259.qm at web90604.mail.mud.yahoo.com>
> Content-Type: text/plain; charset="us-ascii"
>
> 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 at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
> ----- Original Message ----
> From: Brian West <brian.west at mac.com>
> To: freeswitch-users at 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 at 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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