[Freeswitch-users] VOIP vs PSTN

[Michael Jerris]

On Oct 10, 2008, at 10:54 AM, Alfred Richmond wrote:

> 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)
>
> 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>


With just this information, my guess is your call is not hitting the  
same dialplan context.  Turn the log output up to debug and see what  
it is saying, my guess is its falling off the end of the public  
context with no matching extension.

Mike

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