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alfredrichmond at gmai... Guest
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Posted: Fri Oct 10, 2008 10:36 am Post subject: [Freeswitch-users] VOIP vs PSTN |
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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/[url=http://sip.startec.com/14431112222&playback(/usr/local/freeswitch/sounds/warning.wav)]sip.startec.com/14431112222 &playback(/usr/local/freeswitch/sounds/warning.wav)[/url]
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/$1@xx.xx.xx.xx:5061"/>
</condition>
</extension> |
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mike at jerris.com Guest
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Posted: Fri Oct 10, 2008 11:08 am Post subject: [Freeswitch-users] VOIP vs PSTN |
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On Oct 10, 2008, at 10:54 AM, Alfred Richmond wrote:
Quote: | Hello,
I am attempting to generate a message to convert to speech and send it out to my users. I am a newbie but I am just not getting it after reading through the documentation. In testing it works fine when sending to my voip connected users using the following:
bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/internal/1001 &playback(/usr/local/freeswitch/sounds/warning.wav)
however, when I dial a cell phone as below it rings the phone but immediately hangs up before playing the message. Is there something obvious I am doing wrong?
bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/[url=http://sip.startec.com/14431112222&playback(/usr/local/freeswitch/sounds/warning.wav)]sip.startec.com/14431112222 &playback(/usr/local/freeswitch/sounds/warning.wav)[/url]
and then the follow up question is do I need bridge the call in the dialplan like so?
<!-- Dial 11 digit number via startec -->
<extension name="outbound">
<condition field="destination_number" expression="^(\d{11})$">
<!--<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@xx.xx.xx.xx:5061"/>
</condition>
</extension>
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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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