On Fri, Mar 28, 2008 at 9:30 PM, < [EMAIL PROTECTED]> wrote:
> Send Freeswitch-users mailing list submissions to > [email protected] > > 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 > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > 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 <[EMAIL PROTECTED]> > Subject: Re: [Freeswitch-users] New to FreeSwitch > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > 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 <[EMAIL PROTECTED]> > Subject: Re: [Freeswitch-users] How to bridge 2 sessions with > Javascript? > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > 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 > > [email protected] > > 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 <[EMAIL PROTECTED]> > Subject: Re: [Freeswitch-users] How to bridge 2 sessions with > Javascript? > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > 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:[EMAIL PROTECTED] > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED] > pstn:213-799-1400 > > > ----- Original Message ---- > From: Brian West <[EMAIL PROTECTED]> > To: [email protected] > 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 > > [email protected] > > 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 > [email protected] > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080328/d5c4f17f/attachment-0001.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 21, Issue 75 > ************************************************ >
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