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
mod_sofia will now examine a variable in the channel to
see what the channel's originator was using for a codec and
try to put that to the top of the list in the sdp.
if this new sofia profile param is set:
<param name="disable-transcoding" value="true"/>
All outbound calls will use *only* the codec that thier originator
is using to ensure no transcoding.
(of course that could lead to a failed call where there is no way to do this, so use sparingly)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4073 d0543943-73ff-0310-b7d9-9358b9ac24b2
1) The xml_curl now has a more enterprise config where it can have more than 1
url configured so you could have failover. (*note the syntax change*)
2) dialplan modules now take an extra arguement making it possible to pass runtime params to
them. This is now used in mod_dialplan_xml to allow an alternate file path to be specified.
dialplans were already stackable meaning you can configure a sofia profile, for example,
to use enum followed by the default XML dialplan.
e.g. <param name="dialplan" value="enum,XML"/>
From now on, you can also specify :param after each dialplan name to allow param
to be passed to the module. mod_dialplan_xml uses this param as a way to override
where it looks for the dialplan making it possible to stack mutiple calls to the XML dialplan.
e.g. <param name="dialplan" value="XML:/some/xml/file.xml,XML"/>
With this you can search the local file file.xml first and if there is still no match
the hunt will move on to the standard XML using the onboard XML registry and or the external
gateways.
*NOTE* this alternate path does not use the external bindings but it does parse the #includes etc.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4066 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 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
set this sometime before an origination (bridge etc).
<action application="set" data="propagate_vars=my_cool_var1,my_cool_var2,foo,bar"/>
and they should be cloned over to the new channel when it's substantiated
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3101 d0543943-73ff-0310-b7d9-9358b9ac24b2
the 'data' field in action tags may now refer to variables and api functions
to be expanded at runtime.
Syntax:
$varname
${varname}
&func_name(func args)
Exception:
variables that are numeric are still expanded at dialplan compile time based on the regex eg $1 $2 etc
Example:
<extension name="1000">
<condition field="destination_number" expression="^(1000)$">
<action appplication="my_route_app" data="$1"/>
<action appplication="bridge" data="$destination"/>
</condition>
</extension>
Here the $1 is ecaluated before the call begins setting it to 1000 based on the regex ^(1000)$
$destination is evaluated on the fly in execution once the my_route_app has run and has had a
chance to set the variable 'destination' to the correct value.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2994 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
adding mod_park for putting channels in limbo state for remote control.
adding stuff to mod_event_socket to let you do the bgapi <command> <args>
this will let you execute a job in the bg and the result will be sent as an event with an
indicated uuid to match the reply to the command
adding switch_core_port_allocator (to be used soon)
adding "make sure" to do a full rebild of the freeswitch object files
There will be more to this committed as the week progresses
make sure you do a rebuild after this update or you'll be sowwie
./configure && make sure
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2540 d0543943-73ff-0310-b7d9-9358b9ac24b2
codecs need more work...
Expose events into javascript so you can create and fire events and
and pick up chat events with chat-enabled clients like googletalk
EXAMPLE:
session.answer();
e = new Event("custom", "JS::Custom");
e.addHeader("subject", "cool");
e.addBody("hello this is a test");
e.fire;
while(session.ready()) {
session.execute("sleep", "1000");
event = session.getEvent();
if (event) {
str = event.serialize(); // or ("xml")
console_log("debug", "Dump Event:\n" + str + "\n");
e = new Event("custom", "JS::Chat");
e.addHeader("works", "yes");
e.addBody("you said: " + event.getBody());
session.sendEvent(e);
event.destroy();
}
}
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2323 d0543943-73ff-0310-b7d9-9358b9ac24b2
Anthony Minessale