[Freeswitch-users] VOIP vs PSTN

Alfred Richmond alfredrichmond at gmail.com
Fri Oct 10 07:54:06 PDT 2008

I am attempting to generate a message to convert to speech and send it out
to my users. I am a newbie but I am just not getting it after reading
through the documentation. In testing it works fine when sending to my voip
connected users using the following:

bgapi originate

however, when I dial a cell phone as below it rings the phone but
immediately hangs up before playing the message. Is there something obvious
I am doing wrong?

bgapi originate

and then the follow up question is do I need bridge the call in the dialplan
like so?

 <!-- Dial 11 digit number via startec -->
    <extension name="outbound">
     <condition field="destination_number" expression="^(\d{11})$">
      <!--<action application="set"
      <!-- <action application="answer"/>
      <action application="playback"
      <!--<action application="speak" data="cepstral|david|Please hold this
is a test"/> -->
      <action application="bridge" data="sofia/gateway/
sip.startec.com/$1 at xx.xx.xx.xx:5061"/>
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