USAGE <call url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]<br><br>in place of &playback(foo) you can put an extension and optionally a dialplan name and context <br>
and the call will hit the xml dialplan with the extension string as the destination_number field.<br><br><br><br><div class="gmail_quote">On Mon, Apr 7, 2008 at 12:21 PM, <<a href="mailto:tuhl@ix.netcom.com">tuhl@ix.netcom.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I've been using FS for almost a year now as a call-blaster type of<br>
device (originating SIP calls and playing wav/mp3 when user answers).<br>
In the past I was simply sending all calls to a single SIP switch,<br>
but now I need to think about routing to several different SIP<br>
switches, based on NPA-NXX for example.<br>
<br>
The XML dialplan would seem perfect for this, but that's only<br>
triggered by an incoming call, right? I am not sure how to cause<br>
'outbound-only' calls to go through it (since Freeswitch is<br>
originating the call, it only has 1 leg, if that makes sense).<br>
Currently I am originating via XML-RPC from a Perl script, using a<br>
command like 'originate <sip address> &playback(wav file)'. The final<br>
caveat is that we not use javascript to originate the call because I<br>
am still having problems with garbage collection during capacity<br>
testing (the box is pushing out 20cps, 30-second calls, 600<br>
simultaneous calls).<br>
<br>
Thanks,<br>
Tom<br>
<br>
<br>
===============<br>
<a href="mailto:tuhl@ix.netcom.com">tuhl@ix.netcom.com</a><br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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