[Freeswitch-users] Freeswitch-users Digest, Vol 21, Issue 76
Ritesh Singh
riteshsingh81 at gmail.com
Mon Mar 31 06:35:10 PDT 2008
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 at lists.freeswitch.org> wrote:
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> 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 at gmail.com>
> Subject: Re: [Freeswitch-users] New to FreeSwitch (Ritesh Singh)
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> <98a09a460803280948q56f005f6w129adcc4fb6f941a at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Fri, Mar 28, 2008 at 9:30 PM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
> > Send Freeswitch-users mailing list submissions to
> > freeswitch-users at 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 at lists.freeswitch.org
> >
> > You can reach the person managing the list at
> > freeswitch-users-owner at 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 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
> >
> >
> >
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:
> >
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080328/e0635081/attachment-0001.html
> >
> > ------------------------------
> >
> > 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
> >
> >
> > _______________________________________________
> > 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
> >
> >
> >
> >
> >
> >
> >
> >
> ____________________________________________________________________________________
> > Never miss a thing. Make Yahoo your home page.
> > http://www.yahoo.com/r/hs
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> > ------------------------------
> >
> > _______________________________________________
> > 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
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> >
> >
> > End of Freeswitch-users Digest, Vol 21, Issue 75
> > ************************************************
> >
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> ------------------------------
>
> Message: 2
> Date: Fri, 28 Mar 2008 12:01:15 -0700
> From: "Michael Collins" <mcollins at fcnetwork.com>
> Subject: Re: [Freeswitch-users] New to FreeSwitch
> To: <freeswitch-users at lists.freeswitch.org>
> Message-ID:
> <251B739C905BF64FBACCF028D963A5FB021B367C at exchange.fcnetwork.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Ritesh,
>
>
>
> Welcome to the FS community! Brian already mentioned a few things, like
> the wiki and the IRC channel, so definitely get familiar with those.
>
>
>
> One thing I'd like to recommend is that you check out the dialplan xml
> files. (I hope you're comfortable with XML!) In the freeswitch conf
> directory look under dialplan for "default.xml" - this has the pre-built
> dialplan. (You did do "make samples" after the initial install, didn't
> you? :-)) Freeswtich uses regular expressions for matching various
> values in the dialplan. (I hope you like regular expressions! :-))
> Here's an example right from the sample dialplan:
>
>
>
> <extension name="echo">
>
> <condition field="destination_number" expression="^9996$">
>
> <action application="answer"/>
>
> <action application="send_display" data="Echo Test"/>
>
> <action application="echo"/>
>
> </condition>
>
> </extension>
>
>
>
> This is the sample echo test extension. XML helps make some of the
> features obvious, like the extension name. The most interesting thing
> here is the "condition" tag. That line essential says, "If the
> destination number is exactly 9996 then execute these actions." The
> actions are "answer" - that answer the call, then display "Echo Test" on
> the display, and lastly launch the echo application. If you want to
> hear the echo test then try this:
>
> pa call 9996
>
>
>
> If you get the echo test then you know the basic setup of the dial plan
> and port audio is working. Please try it out and report back to the
> list or to IRC. (BTW, do you have an IRC screen name?) We'll go from
> there!
>
>
>
> -MC (IRC: mercutioviz)
>
>
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Ritesh Singh
> Sent: Friday, March 28, 2008 6:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] New to FreeSwitch
>
>
>
> Hi All,
>
> I am very new to freeswitch. It will be great if some one can tell me
> few things:
>
> 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]".
>
> 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.
>
> 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.
>
> This is my second mail as no one replied my first mail. Please do reply
> this mail else i ll be broken ....
>
> Thanks and Regards
> Ritesh
>
>
>
>
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>
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>
> End of Freeswitch-users Digest, Vol 21, Issue 76
> ************************************************
>
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