[Freeswitch-users] VOIP vs PSTN

[Alfred Richmond]

Hello,
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
{ignore_early_media=true,originate_timeout=60}sofia/internal/1001
&playback(/usr/local/freeswitch/sounds/warning.wav)

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
{ignore_early_media=true,originate_timeout=60}sofia/gateway/sip.startec.com/14431112222
&playback(/usr/local/freeswitch/sounds/warning.wav)<http://sip.startec.com/14431112222&playback(/usr/local/freeswitch/sounds/warning.wav)>

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"
data="effective_caller_id_number=4439951026"/>-->
      <!-- <action application="answer"/>
      <action application="playback"
data="/usr/local/freeswitch/sounds/warning.wav"/>
        -->
      <!--<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"/>
     </condition>
    </extension>
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