This adds the | to the originate syntax
making it possible to put a list of urls to call and attempt
to call them one at a time until one of them is successful or there
are none of them left
The original & delimited list is valid for each step in the | separated
list
Example
sofia/test-int/3920@10.3.3.104|sofia/test-int/3910@10.3.3.104&sofia/test-int/3920@10.3.3.104
first call 1 location and if that results in a failure, try 2 at once on the next go
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2996 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
I am not adding it to the examples or to the modules.conf because it's not really ready for that yet.
This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following:
Add this to modules.conf:
-----------------------------------------------------------------------------
endpoints/mod_sofia
-----------------------------------------------------------------------------
Add this to freeswitch.xml in the configuration/modules.conf area
-----------------------------------------------------------------------------
<load module="mod_sofia"/>
-----------------------------------------------------------------------------
Add this to freeswitch.xml in the configuration section
-----------------------------------------------------------------------------
<configuration name="sofia.conf" description="sofia Endpoint">
<!-- You may have multiple profiles -->
<profile name="test">
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="PCMU"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-ip" value="127.0.0.1"/>
<param name="sip-ip" value="127.0.0.1"/>
<!-- optional ; -->
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value="sip:208.64.200.40:5555"/>-->
</profile>
</configuration>
-----------------------------------------------------------------------------
The call string to use profile test would be:
sofia/test/1000@1.2.3.4
as in:
<action application="bridge" data="sofia/test/1000@1.2.3.4"/>
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 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
Anthony Minessale