Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] VOIP vs PSTN


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
alfredrichmond at gmai...
Guest





PostPosted: Fri Oct 10, 2008 10:36 am    Post subject: [Freeswitch-users] VOIP vs PSTN Reply with 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>
Back to top
mike at jerris.com
Guest





PostPosted: Fri Oct 10, 2008 11:08 am    Post subject: [Freeswitch-users] VOIP vs PSTN Reply with quote

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>





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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services