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

[Dave Horton]

I need to do the following:

(1) connect an inbound caller to ivr, and collect a phone number
(2) bridge the caller to the entered phone number
(3) just after connection, play a prompt to the called party 'this call is being recorded'
(4) make a recording of the call

I basically have everything working except number 3 -- playing the prompt to the called party immediately after they pick up. Since I am successfully recording the call I assume we have a conference going on freeswitch, and I should be able to play the prompt to the conference somehow (it's ok if the calling party hears it as well).

I'm doing this in lua, but I think if someone points me to the freeswitch commands to use to stream a file to both parties immediately after bridging that's all I need.

[Abaci]

try api_on_answer or execute_on answer
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
http://wiki.freeswitch.org/wiki/Variable_execute_on_answer

Post by Dave Horton
(1) connect an inbound caller to ivr, and collect a phone number
(2) bridge the caller to the entered phone number
(3) just after connection, play a prompt to the called party 'this call is being recorded'
(4) make a recording of the call
I basically have everything working except number 3 -- playing the prompt to the called party immediately after they pick up. Since I am successfully recording the call I assume we have a conference going on freeswitch, and I should be able to play the prompt to the conference somehow (it's ok if the calling party hears it as well).
I'm doing this in lua, but I think if someone points me to the freeswitch commands to use to stream a file to both parties immediately after bridging that's all I need.
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
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[Avi Marcus]

Set api_on_answer to execute
uuid_broadcast<http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast>
which
will let you pick which legs to play the recording to.
You can pass in the UUID with ${uuid}

-Avi

Post by Abaci
try api_on_answer or execute_on answer
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
http://wiki.freeswitch.org/wiki/Variable_execute_on_answer

Post by Dave Horton
(1) connect an inbound caller to ivr, and collect a phone number
(2) bridge the caller to the entered phone number
(3) just after connection, play a prompt to the called party 'this call

is being recorded'

Post by Dave Horton
(4) make a recording of the call
I basically have everything working except number 3 -- playing the

prompt to the called party immediately after they pick up. Since I am
successfully recording the call I assume we have a conference going on
freeswitch, and I should be able to play the prompt to the conference
somehow (it's ok if the calling party hears it as well).

Post by Dave Horton
I'm doing this in lua, but I think if someone points me to the

freeswitch commands to use to stream a file to both parties immediately
after bridging that's all I need.

Post by Dave Horton
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
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[Dave Horton]

Post by Avi Marcus
Set api_on_answer to execute
uuid_broadcast<
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
which
will let you pick which legs to play the recording to.
You can pass in the UUID with ${uuid}
-Avi

I tried this, and it worked.....sort of. If I play the prompt the A leg it is fine -- caller hears the prompt, then is connected to called party. If I play it to the B leg, however, the prompt does not seem to start playing at all until the B party actually starts talking, and then the prompt is played extremely choppy and slow.....it takes about 10 seconds to play a 4 second prompt. Meanwhile, the caller and called party can't hear each other. Any idea what is going on and how to fix it? I am doing it in lua as follows:

session:execute("export","nolocal:jitterbuffer_msec=100") ;
session:setVariable("RECORD_STEREO","true") ;
session:setVariable("aleg_uuid",uuid) ;
session:execute("record_session", outfile) ;
session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
session:execute("bridge","sofia/normal_customer/${outdial}@${egress_gateway}") ;
--session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid .. " " .. this_call_is_being_recorded .. " " .. "bleg'}sofia/normal_customer/${outdial}@${egress_gateway}") ;

[Michael Collins]

You could try doing a bridge_pre_execute and play the sounds to each leg
independently prior to the bridge. Check out these chan vars:
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_aleg_app
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_aleg_data
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_data

-MC

Post by Dave Horton

Post by Avi Marcus
Set api_on_answer to execute
uuid_broadcast<
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
which
will let you pick which legs to play the recording to.
You can pass in the UUID with ${uuid}
-Avi

I tried this, and it worked.....sort of. If I play the prompt the A leg
it is fine -- caller hears the prompt, then is connected to called party.
If I play it to the B leg, however, the prompt does not seem to start
playing at all until the B party actually starts talking, and then the
prompt is played extremely choppy and slow.....it takes about 10 seconds to
play a 4 second prompt. Meanwhile, the caller and called party can't hear
each other. Any idea what is going on and how to fix it? I am doing it in
session:execute("export","nolocal:jitterbuffer_msec=100") ;
session:setVariable("RECORD_STEREO","true") ;
session:setVariable("aleg_uuid",uuid) ;
session:execute("record_session", outfile) ;
session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
--session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid .. "
" .. this_call_is_being_recorded .. " " ..
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Dave Horton]

I did try that suggestion (api_on_answer uuidbroadcast) and it worked
fine if I played it to the a leg, but when I tried playing to the b leg
the prompt was incredibly choppy and slow, and delayed the cutthru of a
and b legs by a long time. Interestingly, the prompt doesn't even
start playing to the b party until some RTP is received from b for some
reason, and then it sort of slowly chuffs out the prompt in small
choppy chunks.

I then tried Michael's follow on suggestion of using
bridge_pre_execute_bleg_app, but that didn't work at all -- no prompt
heard.

I then decided to try the outdial as a bridged conference, and then
play a prompt to the conference after the b party picks up. This may
actually be the best option for me as I need the prompt to be saved in
the recording of the call, and I am guessing when I stream it out on a
bridged call it may not show up in the recording.

However, I had no luck getting a bridged conference outdial to work at
all -- the outdial does work successfully, but a and b parties don't
hear each other so one must not be in the conference. And then when
the b party hangs up the a party is not automatically hung up, which
should happen in a bridged conference if I understand things correctly.

So, while I am still interested in what is causing the prompt on the
bridge call to sound so bad when played to the b leg, perhaps what is
more important to me now is getting the bridged conference to work.
Here is what I am doing, in case anyone can point out the problem:

session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
session:execute("conference","bridge:_uuid_ at simple:sofia/normal_customer/${outdial}@${egress_gateway}")
;
session:setVariable("conference_enter_sound",this_call_is_being_recorded) ;

(I created a conference profile called 'simple' that turns off caller
controls; interestingly, that doesn't seem to be picked up either as
the debug logging shows it attaching the default caller controls)

Dave

Post by Dave Horton

Post by Avi Marcus
Set api_on_answer to execute
uuid_broadcast<
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
which
will let you pick which legs to play the recording to.
You can pass in the UUID with ${uuid}
-Avi

I tried this, and it worked.....sort of. If I play the prompt the A
leg it is fine -- caller hears the prompt, then is connected to called
party. If I play it to the B leg, however, the prompt does not seem to
start playing at all until the B party actually starts talking, and
then the prompt is played extremely choppy and slow.....it takes about
10 seconds to play a 4 second prompt. Meanwhile, the caller and called
party can't hear each other. Any idea what is going on and how to fix
session:execute("export","nolocal:jitterbuffer_msec=100") ;
session:setVariable("RECORD_STEREO","true") ;
session:setVariable("aleg_uuid",uuid) ;
session:execute("record_session", outfile) ;
session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
--session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid ..
" " .. this_call_is_being_recorded .. " " ..
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Michael Collins]

I would try to get this working w/ just raw XML dialplan and take Lua out
of the equation. Once you get it working then migrate into a Lua script,
assuming you even need to use it. Personally, I try not to overuse Lua (or
any other dp scripting language) and keep as much in the XML dialplan as
possible.

Lab it up just with the dialplan using as simple a config as possible.
Capture logs and post to pastebin.freeswitch.org along with your relevant
configs (dialplan, conference) and some of the gang here will see if they
can help analyze what's going on.

-MC

Post by Dave Horton
I did try that suggestion (api_on_answer uuidbroadcast) and it worked
fine if I played it to the a leg, but when I tried playing to the b leg
the prompt was incredibly choppy and slow, and delayed the cutthru of a
and b legs by a long time. Interestingly, the prompt doesn't even
start playing to the b party until some RTP is received from b for some
reason, and then it sort of slowly chuffs out the prompt in small
choppy chunks.
I then tried Michael's follow on suggestion of using
bridge_pre_execute_bleg_app, but that didn't work at all -- no prompt
heard.
I then decided to try the outdial as a bridged conference, and then
play a prompt to the conference after the b party picks up. This may
actually be the best option for me as I need the prompt to be saved in
the recording of the call, and I am guessing when I stream it out on a
bridged call it may not show up in the recording.
However, I had no luck getting a bridged conference outdial to work at
all -- the outdial does work successfully, but a and b parties don't
hear each other so one must not be in the conference. And then when
the b party hangs up the a party is not automatically hung up, which
should happen in a bridged conference if I understand things correctly.
So, while I am still interested in what is causing the prompt on the
bridge call to sound so bad when played to the b leg, perhaps what is
more important to me now is getting the bridged conference to work.
session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
session:execute("conference","bridge:_uuid_ at simple
;
session:setVariable("conference_enter_sound",this_call_is_being_recorded) ;
(I created a conference profile called 'simple' that turns off caller
controls; interestingly, that doesn't seem to be picked up either as
the debug logging shows it attaching the default caller controls)
Dave

Post by Dave Horton

Post by Avi Marcus
Set api_on_answer to execute
uuid_broadcast<
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
which
will let you pick which legs to play the recording to.
You can pass in the UUID with ${uuid}
-Avi

I tried this, and it worked.....sort of. If I play the prompt the A
leg it is fine -- caller hears the prompt, then is connected to called
party. If I play it to the B leg, however, the prompt does not seem to
start playing at all until the B party actually starts talking, and
then the prompt is played extremely choppy and slow.....it takes about
10 seconds to play a 4 second prompt. Meanwhile, the caller and called
party can't hear each other. Any idea what is going on and how to fix
session:execute("export","nolocal:jitterbuffer_msec=100") ;
session:setVariable("RECORD_STEREO","true") ;
session:setVariable("aleg_uuid",uuid) ;
session:execute("record_session", outfile) ;
session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;

${egress_gateway}")

Post by Dave Horton
;
--session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid ..
" " .. this_call_is_being_recorded .. " " ..
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Dave Horton]

Post by Michael Collins
Lab it up just with the dialplan using as simple a config as possible.
Capture logs and post to pastebin.freeswitch.org along with your
relevant configs (dialplan, conference) and some of the gang here will
see if they can help analyze what's going on.

Ok, I reproduced the scenario where it is a bridged call (not a
conference) where I play the prompt after answer and get the extremely
choppy audio (10 seconds or more to play a 4 second prompt). The
pastebin of the debug logging is at
http://pastebin.freeswitch.org/19785, and the configuration I used is
below. If anyone can point out what might be causing this it would be
appreciated. Next I will try the conferencing version and paste those
results

<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="export"
data="nolocal:api_on_answer=uuid_broadcast ${uuid}
misc/this-call-is-being-recorded.wav bleg"/>
<action application="bridge"
data="sofia/normal_customer/15083084809@${egress_gateway}"/>

</condition>
</extension>

[Dave Horton]

Post by Dave Horton
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>

[Michael Collins]

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

Post by Dave Horton

Post by Dave Horton
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>
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Dave Horton]

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

Post by Michael Collins
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

Post by Dave Horton
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"
? ? ? ? ? ? ? ? <action application="set"
data="conference_enter_sound=misc/this-call-is-being-recorded"/>
? ? ? ? </condition>
? ? </extension>
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Michael S Collins
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
_________________________________________________________________________
consulting at freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org