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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ritesh Singh Sent: Friday, March 28, 2008 6:08 AM To: freeswitch-users@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