[Freeswitch-users] how to play a prompt after connection

[Dave Horton]

OK, here is a pastebin minus the sip tracing with the highlighting: 
http://pastebin.freeswitch.org/19788

On 2012-08-30 19:39:02 +0000, Michael Collins said:

> Dave,
> Can you turn off the sofia debugging? It's almost completely useless in 
> these scenarios. (only a few devs really understand it and they'll let 
> you know if you need to turn it on.) Also, please use "FreeSWITCH Log" 
> as syntax highlighting - that makes life much easier for those of us 
> whose eyes have higher mileage...
> -MC
> 
> On Thu, Aug 30, 2012 at 12:31 PM, Dave Horton 
> <daveh at beachdognet.com> wrote:
> On 2012-08-30 18:59:34 +0000, Dave Horton said:
> 
> > Next I will try the conferencing version and paste those
> > results
> 
> Below is the configuration I used with the bridged conferencing.
> Something must be wrong because there is no audio path between the a
> and b legs; also, when b hangs up a stays connected.
> 
> Pastebin is at http://pastebin.freeswitch.org/19786 and configuration
> is below.  I'd welcome any thoughts on what is wrong...
> 
> 
>     <extension name="6174973896" continue="false">
>       <condition field="source" expression="mod_sofia"/>
>       <condition field="destination_number" expression="6174973896">
>                 <action application="set" data="call_limit=1"/>
>                 <action application="set" data="callFactor=1"/>
>                 <action application="set" data="customer=cps_play"/>
>                 <action application="ring_ready"/>
>                 <action application="pre_answer"/>
>                 <action application="start_dtmf"/>
>                 <action application="set"
> data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
>                 <action application="set"
> data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
>                 <action application="export"
> data="nolocal:jitterbuffer_msec=100"/>
>                 <action application="set" data="RECORD_STEREO=true"/>
>                 <action application="set" data="aleg_uuid=${uuid}"/>
>                 <action application="set"
> data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
>                 <action application="set"
> data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
>                 <action application="record_session" data="${outfile}"/>
>                 <action application="set" data="progress_timeout=60"/>
>                 <action application="set" data="call_timeout=120"/>
>                 <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
>                 <action application="set" data="hangup_after_bridge=true"/>
>                 <action application="conference"
> data="bridge:_uuid_ at simple:sofia/normal_customer/15083084809@${egress_gateway}"/> 
> 
> 
>                 <action application="set"
> data="conference_enter_sound=misc/this-call-is-being-recorded"/>
> 
>         </condition>
>     </extension>
> 
> 
> 
> _________________________________________________________________________
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> 
> 
> 
> 
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> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
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