1. 3 retries for pin entry
2. don't require pin entry for outbound calls
3. allow outbound calls to enter the conference when locked
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3553 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$">
<action application="set" data="RECORD_TITLE=recording test"/>
<action application="set" data="RECORD_ARTIST=FreeSWITCH"/>
<action application="record_session" data="/tmp/rtest.wav"/>
<action application="playback" data="/tmp/some.wav"/>
<action application="stop_record_session" data="/tmp/rtest.wav"/>
<action application="playback" data="/tmp/some_other.wav"/>
</condition>
</extension>
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2588 d0543943-73ff-0310-b7d9-9358b9ac24b2
no more <rooms>
rooms will just be created on the fly.
<action application="conference" data="myconf@myprofile+1234"/>
+1234 is optional on-the-fly pin
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2381 d0543943-73ff-0310-b7d9-9358b9ac24b2
Ok,
This one adds a bunch of stuff on top of the framework restructuring from yesterday.
1) originate api function:
Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
This will call the specified url then transfer the call to the specified extension
example: originate exosip/1000@somehost 1000 XML default
2) mutiple destinations in outbound calls:
This means any dialstring may contain an '&' separated list of call urls
When using mutiple urls in this manner it is possible to map a certian key as required
indication of an accepted call. You may also supply a filename to play possibly instructing the
call recipiant to press the desired key etc...
The example below will call 2 locations playing prompt.wav to any who answer and
completing the call to the first offhook recipiant to dial "4"
<extension name="3002">
<condition field="destination_number" expression="^3002$">
<action application="set" data="call_timeout=60"/>
<action application="set" data="group_confirm_file=/path/to/prompt.wav"/>
<action application="set" data="group_confirm_key=4"/>
<action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/>
</condition>
</extension>
The following is the equivilant but the confirm data is passed vial the bridge parameters
(This is for situations where there is no originating channel to set variables to)
<extension name="3002">
<condition field="destination_number" expression="^3002$">
<action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/>
</condition>
</extension>
Omitting the file and key stuff will simply comeplete the call to whoever answers first.
(this is similar to how other less fortunate software handles the situation with thier best effort.)
This logic should be permitted in anything that establishes an outgoing call with
switch_ivr_originate()
Yes! That means even in this new originate api command you can call mutiple targets and send
whoever answers first to an extension that calls more mutiple targets. (better test it though!)
Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features
please report any behaviour contrary to this account to me ASAP cos i would not be terribly
suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
