<div>I am not sure what exactly you wish to achieve. Just a basic SIP--to--SIP call or ?</div> <div> </div> <div>I am not much into the configs, but ya I can tell you that you can try using FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then u editing them, as it has macros, context etc... which is too high to me. But the browser interface help a lot understanding the config files later once configured via FreePBX.</div> <div> </div> <div>FreePBX -- Its a tool(software which is wrapper over asterisk which gives a web based interface to manage & configure ur asterisk configuration files with easy understanding.</div> <div> </div> <div>tixbox-- Its a kind of Asterisk solution which is combination of asterisk+freepbx+linux+crm tools etc.. for quick Asterisk deployment.</div> <div> </div> <div>I am not sure whether u know all these if yes, hen excuse me.. but ur mail sounded u might need this info
needed.<BR><BR><B><I>sanchal.singh@alliance-infotech.com</I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Hi,<BR>I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running<BR><BR>1st PC is having IP Adress : 192.168.1.149<BR>2nd PC is having IP Adress : 192.168.1.53<BR><BR>Now, I am tying to dial from 1st PC to 2nd PC<BR><BR>I am trying to dial from 1st PC to 2nd PC through asterisk server<BR>The problem is 1st PC is calling directly to 2nd PC not through asterisk server<BR><BR>I am doing the following additions in configuration files<BR><BR>1) sip.conf<BR><BR>[general]<BR>context=sip<BR>bindport=5060 <BR>bindaddr=0.0.0.0
<BR><BR>[phone1]<BR>type=friend<BR>host=192.168.1.149<BR>port=5060<BR>nat=yes<BR>dtmfmode=rfc2833<BR>context=sip<BR><BR>[phone2]<BR>type=friend<BR>host=192.168.1.53<BR>port=5060<BR>nat=yes<BR>dtmfmode=rfc2833<BR>context=sip<BR><BR>2) extensions.conf<BR>exten => 11,1,Dial(SIP/phone2,20,tr)<BR><BR>Now, I am calling from sip phone1 by name phone2@192.168.1.53<BR>It is not being called through asterisk server running on linux m/c. It is calling directly. As, I am running "sip debub" but no packet dumping is taking place. Can anybody will tell me the error I am doing.<BR>Thanx and regards<BR>sanchal<BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR><BR>_______________________________________________<BR>--Bandwidth and Colocation Provided by http://www.api-digital.com--<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><BR><BR><BR><DIV>
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