On Fri, Mar 28, 2008 at 9:30 PM, <
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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@mac.com>
Subject: Re: [Freeswitch-users] New to FreeSwitch
To: freeswitch-users@lists.freeswitch.org
Message-ID: <CE2A83CB-29B3-41B9-BD01-DFD7861AC801@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@dalethatcher.com>
Subject: Re: [Freeswitch-users] How to bridge 2 sessions with
Javascript?
To: freeswitch-users@lists.freeswitch.org
Message-ID: <1206717129.8070.51.camel@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
>
>
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Message: 3
Date: Fri, 28 Mar 2008 08:16:50 -0700 (PDT)
From: Anthony Minessale <anthmct@yahoo.com>
Subject: Re: [Freeswitch-users] How to bridge 2 sessions with
Javascript?
To: freeswitch-users@lists.freeswitch.org
Message-ID: <539060.7259.qm@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/
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MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
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iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
----- Original Message ----
From: Brian West <brian.west@mac.com>
To: freeswitch-users@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
>
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