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
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 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 modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan.
It adds some API interface calls usable from a remote client such as mod_event_socket or the test console.
1) media [off] <uuid>
Turns on/off the media on the call described by <uuid>
The media will be redirected as desiered either into the switch or point to point.
2) hold [off] <uuid>
Turns on/off endpoint specific hold state on the session described by <uuid>
3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both]
A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated.
If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified
will hear the message.
During playback when only one side is hearing the message the other end will hear silence.
If media is not flowing across the switch when the message is broadcasted, the media will be directed to the
switch for the duration of the call and then returned to it's previous state.
Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session
description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media
on the switch.
<action application="set" data="no_media=true"/>
<action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/>
*NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled,
the media for the first leg will be engaged with the switch until the second leg has answered and the other session description
is available to establish a point to point connection at which time point-to-point mode will be enabled.
*NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
Transfers work better when both legs of the call live in thier own channel eg bridged calls
A -> B where you want a to make B -> C
when you route a call to an IVR or playback app you are not really bridging you have
A all alone executing the script so it's hard to transfer that.
I do have it aparently working but it's goofy and you are better off
putting your IVR on it's own switch so they are all inbound calls
then you have A -> B -> IVR
now A can happily transfer B who can stay on line with IVR without stopping
the execution. You can also accomplish this by calling in a loop back to the same box
if you dont want to have 2 boxes.
Also the beginning effort at bridging calls with no media is here
set this magic variable in your dialplan to convince mod_sofia
to pass A's sdp as it's own to B and return B's sdp back to A on 200 or 183
<action application="set" data="no_media=true"/>
<action application="bridge" data="sofia/id@host.com"/>
You will need a new sofia tarball for this version
There is a bunch of other odds and ends added like a function or 2 etc
Oh,
And don't be suprised if it introduces all kinds of bugs!
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2992 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
Michael Jerris
Anthony Minessale
Mike Murdock