<br><br><div class="gmail_quote">On Fri, Mar 28, 2008 at 9:30 PM,  &lt;<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>&gt; wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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<br>When replying, please edit your Subject line so it is more specific<br>than &quot;Re: Contents of Freeswitch-users digest...&quot;<br><br><br>Today&#39;s Topics:<br><br>&nbsp; 1. Re: New to FreeSwitch (Brian West)<br>&nbsp; 2. Re: How to bridge 2 sessions with Javascript? (Dale Thatcher)<br>
&nbsp; 3. Re: How to bridge 2 sessions with Javascript? (Anthony Minessale)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Fri, 28 Mar 2008 09:57:23 -0500<br>From: Brian West &lt;<a href="mailto:brian.west@mac.com">brian.west@mac.com</a>&gt;<br>
Subject: Re: [Freeswitch-users] New to FreeSwitch<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>Message-ID: &lt;<a href="mailto:CE2A83CB-29B3-41B9-BD01-DFD7861AC801@mac.com">CE2A83CB-29B3-41B9-BD01-DFD7861AC801@mac.com</a>&gt;<br>
Content-Type: text/plain; charset=&quot;us-ascii&quot;<br><br><br>On Mar 28, 2008, at 8:07 AM, Ritesh Singh wrote:<br><br>&gt; Hi All,<br>&gt;<br>&gt; I am very new to freeswitch. It will be great if some one can tell<br>
&gt; me few things:<br><br>Welcome to the community.<br><br>&gt;<br>&gt;<br>&gt; 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y.<br>&gt; Then what changes i am supposed to do and at what place. I would<br>
&gt; like to use the mod_portaudio for this purpose.<br>&gt;<br>&gt; 2) I started the windows freeswitch and loaded port_audio by using<br>&gt; &quot;load mod_portaudio&quot; , then used &quot;pa call 1234&quot;...the session gets<br>
&gt; initiated but the call gets hanged up with the log &quot;portaudio/1234<br>&gt; [CS_RING] [NO_ROUTE_DESTINATION]&quot;.<br><br>I&#39;m sure this message is rather clear... You do not have a route in<br>your dialplan for 1234 thus it fails.<br>
<br>&gt; 3) Suppose i have a jabber server <a href="http://x.net/" target="_blank">x.net</a>. and i have 2 users having<br>&gt; account at that jabber server. Then how can i use mod_dingaling so<br>&gt; that i have the voice chat between those user of the jabber server<br>
&gt; <a href="http://x.net/" target="_blank">x.net</a>.<br>&gt; Also, i would like to dump the all the record regarding the call<br>&gt; made, like duration of call , person who initiated the call, call<br>&gt; destination , etc.<br>
<br>Feels like your trying to do too much at once before you fully<br>understand what is going on.<br><br>&gt; Any help in this regard is highly appreciable. &nbsp;Please &nbsp;do forgive<br>&gt; me if you think its stupid mail but i am desparate &nbsp;for these answers.<br>
&gt;<br><br>You&#39;re trying to fly before you can even crawl. &nbsp;You have many things<br>to learn.<br><br>Best things to do are start here <a href="http://wiki.freeswitch.org/" target="_blank">http://wiki.freeswitch.org</a> and<br>
#freeswitch on <a href="http://irc.freenode.net/" target="_blank">irc.freenode.net</a><br><br>Remember I&#39;ll be very glad to answer questions but the requirement is<br>that you MUST put the info you learn on the wiki and pay it forward to<br>
others.<br><br>Thanks,<br>/b<br><br><br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <a href="http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080328/e0635081/attachment-0001.html" target="_blank">http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080328/e0635081/attachment-0001.html</a><br>
<br>------------------------------<br><br>Message: 2<br>Date: Fri, 28 Mar 2008 15:12:09 +0000<br>From: Dale Thatcher &lt;<a href="mailto:freeswitch@dalethatcher.com">freeswitch@dalethatcher.com</a>&gt;<br>Subject: Re: [Freeswitch-users] How to bridge 2 sessions with<br>
&nbsp; &nbsp; &nbsp; &nbsp;Javascript?<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>Message-ID: &lt;1206717129.8070.51.camel@desktop&gt;<br>Content-Type: text/plain<br><br>Great, guess I&#39;ll drop that from my config then. &nbsp;Sorry Nicolas must be<br>
something else.<br><br>- Dale<br><br>On Fri, 2008-03-28 at 09:53 -0500, Brian West wrote:<br>&gt; Dale,<br>&gt; &nbsp; &nbsp; &nbsp; This was due to a rogue SRV record saying use TCP that has since been<br>&gt; corrected after a month of talking to them about it.<br>
&gt;<br>&gt; /b<br>&gt;<br>&gt; On Mar 28, 2008, at 8:50 AM, Dale Thatcher wrote:<br>&gt;<br>&gt; &gt; I had some slow connection problems with sipphone that were solved by:<br>&gt; &gt;<br>&gt; &gt; &nbsp; &nbsp; &lt;param name=&quot;bind-params&quot; value=&quot;transport=udp&quot;/&gt;<br>
&gt; &gt;<br>&gt; &gt; in the sip profile, might be worth a go. &nbsp;BTW this is a bug with<br>&gt; &gt; sipphone, not with Freeswitch.<br>&gt; &gt;<br>&gt; &gt; - Dale<br>&gt;<br>&gt;<br>&gt; _______________________________________________<br>
&gt; Freeswitch-users mailing list<br>&gt; <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>&gt; <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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------------------------------------------------<br><a href="http://myhelpa.com/" target="_blank">http://myhelpa.com</a> - Sign up to be a Beta Helpa.<br><br><br><br>------------------------------<br><br>Message: 3<br>Date: Fri, 28 Mar 2008 08:16:50 -0700 (PDT)<br>
From: Anthony Minessale &lt;<a href="mailto:anthmct@yahoo.com">anthmct@yahoo.com</a>&gt;<br>Subject: Re: [Freeswitch-users] How to bridge 2 sessions with<br>&nbsp; &nbsp; &nbsp; &nbsp;Javascript?<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Message-ID: &lt;<a href="mailto:539060.7259.qm@web90604.mail.mud.yahoo.com">539060.7259.qm@web90604.mail.mud.yahoo.com</a>&gt;<br>Content-Type: text/plain; charset=&quot;us-ascii&quot;<br><br>and just to point it out, you can also do:<br>
<br>session1 = new Session();<br>session1.originate(session1, &quot;{ignore_early_media=true}sofia/gateway/<br><a href="http://asterlink.com/19184249378" target="_blank">asterlink.com/19184249378</a>&quot;);<br><br>session1.execute(&quot;bridge&quot;, &quot;sofia/gateway/asterlink.com/19184238080&quot;);<br>
<br>or even better<br><br>// this will transfer the channel into the dialplan and end the script<br>// thus reducing overhead of leaving JS open.<br>session1.execute(&quot;transfer&quot;, &quot;19184238080&quot;);<br><br>
<br><br><br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN:anthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/PAYPAL:<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a><br>IRC: <a href="http://irc.freenode.net/" target="_blank">irc.freenode.net</a> #freeswitch<br>
<br>FreeSWITCH Developer Conference<br><a href="mailto:sip:888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk:conf+888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br><br><br>----- Original Message ----<br>From: Brian West &lt;<a href="mailto:brian.west@mac.com">brian.west@mac.com</a>&gt;<br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>Sent: Thursday, March 27, 2008 9:55:25 PM<br>Subject: Re: [Freeswitch-users] How to bridge 2 sessions with Javascript?<br>
<br>Revised script:<br><br>session1 = new Session();<br>session1.originate(session1, &quot;{ignore_early_media=true}sofia/gateway/<br><a href="http://asterlink.com/19184249378" target="_blank">asterlink.com/19184249378</a>&quot;);<br>
<br>session2 = new Session();<br>session2.originate(session2, &quot;sofia/gateway/asterlink.com/19184238080&quot;);<br><br>bridge(session1, session2);<br>session.hangup();<br>while (session1.ready() &amp;&amp; session2.ready()) { }<br>
<br><br>You don&#39;t need waitForAnswer because the ignore_early_media=true<br>performs that automatically for you so it won&#39;t return till you<br>answer. &nbsp;And you don&#39;t want the ignore_early_media on the second leg<br>
otherwise you get NO ringback and its just silent till the other end<br>is answered.<br><br>/b<br><br>On Mar 27, 2008, at 9:10 PM, Nicolas Brenner wrote:<br><br>&gt; Hello everybody again,<br>&gt;<br>&gt; First of all I want to say thanks to the people on this list and on<br>
&gt; IRC, I&#39;m really surprised (in a very good way) of the help I&#39;ve<br>&gt; received.<br>&gt;<br>&gt; Now to my problem: I&#39;m trying to bridge two SIP calls together with a<br>&gt; JS script, and to achieve it, I&#39;m did the following:<br>
&gt; - created a dialplan entry for extension 500 which calls js script<br>&gt; - created a js script with the following code:<br>&gt;<br>&gt; // Create new_session<br>&gt; session1 = new Session();<br>&gt; session1.originate(session,<br>
&gt; &quot;{ignore_early_media=true}sofia/gateway/sip.sipdiscount.com/<br>&gt; 005624949458&quot;);<br>&gt; session1.waitForAnswer(10000);<br>&gt;<br>&gt; new_session = new Session();<br>&gt; new_session.originate(session,<br>
&gt; &quot;{ignore_early_media=true}sofia/gateway/sip.sipdiscount.com/<br>&gt; 0056979039388&quot;,<br>&gt; 30);<br>&gt; new_session.waitForAnswer(10000);<br>&gt;<br>&gt; // IF everybody is ready, then bridge our current session &amp; the<br>
&gt; new_session<br>&gt; if (session1.ready() &amp;&amp; new_session.ready()) {<br>&gt; &nbsp; &nbsp; &nbsp; &nbsp;console_log(&quot;info&quot;, &quot;Interoligofrenico!\n&quot;);<br>&gt; &nbsp; &nbsp;bridge(session1, new_session);<br>&gt; }<br>&gt;<br>
&gt; // hangup when done<br>&gt; session1.hangup();<br>&gt; new_session.hangup();<br>&gt;<br>&gt; When I register with extension 1000 using a softphone and dial<br>&gt; extension 500, the code above successfully creates the two new<br>
&gt; sessions, makes the calls in order, and supposedly bridges the<br>&gt; sessions (I get no error about the bridge on the console, and I get<br>&gt; the log text too), but I get no audio on either end. Anybody know<br>
&gt; what&#39;s wrong with the code?<br>&gt;<br>&gt; Btw, how can I originate a call to a configure extension? All examples<br>&gt; on the wiki use the &#39;sofia syntax&#39;, should I just use<br>&gt; sofia/default/1001 for example?<br>
&gt;<br>&gt; Thanks!<br>&gt;<br>&gt; Nicolas<br>&gt;<br>&gt; _______________________________________________<br>&gt; Freeswitch-users mailing list<br>&gt; <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
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<a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br><br><br><br><br><br><br>&nbsp; &nbsp; &nbsp;____________________________________________________________________________________<br>Never miss a thing. &nbsp;Make Yahoo your home page.<br>
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<a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br><br><br>End of Freeswitch-users Digest, Vol 21, Issue 75<br>************************************************<br></blockquote></div><br>