This patch adds a scheduler thread to the core and moves the heartbeat
event to use the new scheduler as an example.
Also The following features are implemented that use this scheduler:
sched_hangup dialplan application:
<action application="sched_hangup" data="+10 normal_clearing bleg"/>
** The cause code is optional and the optional bleg keyword will only hangup the
channel the current channel is bridged to if the call is in a bridge.
sched_transfer dialplan application:
<action application="sched_transfer" data="+10 1000 XML default"/>
** The last 2 args (dialplan and context) are optional
sched_broadcast dialplan application:
<action application="sched_broadcast" data="+10 playback:/tmp/foo.wav"/>
<action application="sched_broadcast" data="+10 playback!normal_clearing:/tmp/foo.wav"/>
** The optional !<cause_code> can be added to make the channel hangup after broadcasting the file.
sched_hangup api function:
sched_hangup +10 <uuid_string> normal_clearing
** The cause code is optional
sched_transfer api function:
sched_transfer +10 <uuid_string> 1000 XML default
** The last 2 args (dialplan and context) are optional
sched_broadcast api function:
sched_broadcast +10 <uuid_str> playback:/tmp/foo.wav
sched_broadcast +10 <uuid_str> playback!normal_clearing:/tmp/foo.wav
** The optional !<cause_code> can be added to make the channel hangup after broadcasting the file.
The new C functions in the core are documented in the doxeygen.
*NOTE* This commit should satisfy at least 2 bounties on the wiki
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4785 d0543943-73ff-0310-b7d9-9358b9ac24b2
The most important thing to check is you now must create a new session with a blank constructor:
s = new Session();
then call s.originate() with all the former args that were documented to be for the constructor
this will will return true or false to indicate if the call worked.
See below this sample code demonstrates all of the changes:
////////////////////////////////////////////////////////////////////////////////
function on_hangup(hup_session)
{
console_log("debug", "HANGUP!!!! name: " + hup_session.name + " cause: " + hup_session.cause + "\n");
//exit here would end the script so you could cleanup and just be done
//exit();
}
//set the on_hangup function to be called when this session is hungup
session.setHangupHook(on_hangup);
//allocate a new b_session
var b_session = new Session();
//make a call with b_session. If this fails, all we will be able to access is the b_session.cause attr
if (b_session.originate(session, "sofia/mydomain.com/888@conference.freeswitch.org")) {
//Inform the scripting engine to automaticly hang this session up when the script ends
b_session.setAutoHangup(true);
//set the on_hangup function to be called when this session is hungup
b_session.setHangupHook(on_hangup);
//bridge session with b_session
bridge(session, b_session);
} else {
console_log("debug", "Originate Failed.. cause: " + b_session.cause + "\n");
}
////////////////////////////////////////////////////////////////////////////////
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4773 d0543943-73ff-0310-b7d9-9358b9ac24b2
change most char * values in ivr_menu functions to const char *
change switch_core_strdup to get passed const char * instead of char *
change switch_xml_find_child to get passed const char * instead of char *
change the ivr dialplan application to free the xml config as soon as it is done building the xml menu and not hold it until the menu is done being run, so that you can do a reloadxml while someone is in a menu without blocking.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4260 d0543943-73ff-0310-b7d9-9358b9ac24b2
This addition lets you set artifical ringback on a channel
that is waiting for an originated call to be answered.
the syntax is
<action application="set" data="ringback=[data]"/>
where data is either the full path to an audio file
or a teletone generation script..
syntax of teletone scripts
LEGEND:
0-9,a-d,*,# (standard dtmf tones)
variables: c,r,d,v,>,<,+,w,l,L,%
c (channels) - Sets the number of channels.
r (rate) - Sets the sample rate.
d (duration) - Sets the default tone duration.
v (volume) - Sets the default volume.
> (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration).
< (increase vol) - factor to increase volume by per frame (0 for even increase across duration).
+ (step) - factor to step by used by < and >.
w (wait) - default silence after each tone.
l (loops) - number of times to repeat each tone in the script.
L (LOOPS) - number of times to repeat the the whole script.
% (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies.
standard tones can have custom duration per use with the () modifier
7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds
EXAMPLES
UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms]
%(400,200,400,450);%(400,2200,400,450)
US Ring Tone [440+480 hz on for 2000ms off for 4000ms]
%(2000,4000,440,480)
ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms
v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)
SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait
%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)
ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait
%(100,100,1400,2060,2450,2600)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
This changes the core to have the necessary tools to create
a speech detection interface.
It also changes the code in javascript (mod_spidermonkey)
there are a few api changes in how it handles callbacks
It also adds grammars as a system dir to store asr grammars
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3291 d0543943-73ff-0310-b7d9-9358b9ac24b2
This is the primary commit to add bugs to the core (media bugs that is)
Media bugs are kind of like what ChanSpy is in Asterisk only cooler (I wrote ChanSpy too so I can say that)
Here is an example of using them to record a call by the higher level switch_ivr functionality passed
up to the dialplan via mod_playback.
The call will be recorded while the some.wav plays then stop for the rest of the call (when some_other.wav plays)
The bugs may have bugs since this is 1 day's work so happy hunting ......
<extension name="42">
<condition field="destination_number" expression="^42$">
<action application="set" data="RECORD_TITLE=recording test"/>
<action application="set" data="RECORD_ARTIST=FreeSWITCH"/>
<action application="record_session" data="/tmp/rtest.wav"/>
<action application="playback" data="/tmp/some.wav"/>
<action application="stop_record_session" data="/tmp/rtest.wav"/>
<action application="playback" data="/tmp/some_other.wav"/>
</condition>
</extension>
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2588 d0543943-73ff-0310-b7d9-9358b9ac24b2
