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
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
now in the key portion you can say 'exec' and in the file portion say '<application> <args>'
if the channel is not hungup when that application ends it's the winner so you can
run an ivr on the channels to determine who gets the call
<extension name="3002">
<condition field="destination_number" expression="^3002$">
<action application="set" data="call_timeout=60"/>
<action application="set" data="group_confirm_key=exec"/>
<action application="set" data="group_confirm_file=javascript test.js"/>
<action application="bridge" data="exosip/1000@domain.com&exosip/1001@mydomain.com"/>
</condition>
</extension>
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2314 d0543943-73ff-0310-b7d9-9358b9ac24b2
Ok,
This one adds a bunch of stuff on top of the framework restructuring from yesterday.
1) originate api function:
Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
This will call the specified url then transfer the call to the specified extension
example: originate exosip/1000@somehost 1000 XML default
2) mutiple destinations in outbound calls:
This means any dialstring may contain an '&' separated list of call urls
When using mutiple urls in this manner it is possible to map a certian key as required
indication of an accepted call. You may also supply a filename to play possibly instructing the
call recipiant to press the desired key etc...
The example below will call 2 locations playing prompt.wav to any who answer and
completing the call to the first offhook recipiant to dial "4"
<extension name="3002">
<condition field="destination_number" expression="^3002$">
<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
