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
This is some brand new stuff to gateway chat/presence/audio from one protocol to another
So far it only works between google/jingle and SIP
All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end.
With this setup registered X-Lite's can chat with each other and call each other
as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls.
Chat May also be done between X-Lite and jabber
You'll also need a jabber server configured for component login so you can interface.
We have only tested with jabberd2 so far.
Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example)
so the jabber records are pointed at your jabber server.
RELEVANT CONFIGS
<!-- Brian has no jingle support so send calls to him over to his iax url -->
<extension name="bkw">
<condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$">
<action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/>
</condition>
</extension>
<!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below -->
<extension name="jingle2sip">
<condition field="source" expression="mod_dingaling"/>
<condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$">
<action application="bridge" data="sofia/$2/$1%$2"/>
</condition>
</extension>
<extension name="sip2jingle">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$">
<action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/>
</condition>
</extension>
<configuration name="sofia.conf" description="sofia Endpoint">
<global_settings>
<param name="log-level" value="0"/>
</global_settings>
<profiles>
<profile name="fs.mydomain.com">
<registrations/>
<settings>
<param name="debug" value="1"/>
<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="codec-ms" value="20"/>
<param name="accept-blind-reg" value="true"/>
<param name="manage-presence" value="true"/>
<!--<param name="full-id-in-dialplan" value="true"/>-->
<!--<param name="auth-calls" value="true"/>-->
<!--<param name="auth-all-packets" value="true"/>-->
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="100.200.100.200"/>
<param name="sip-ip" value="fs.mydomain.com"/>
</settings>
</profile>
</profiles>
</configuration>
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<profile type="component">
<param name="name" value="fs.mydomain.com"/>
<param name="password" value="secret"/>
<param name="dialplan" value="XML"/>
<param name="rtp-ip" value="208.64.200.42"/>
<param name="server" value="jabber.freeswitch.org:5347"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<param name="exten" value="_auto_"/>
<!--<param name="vad" value="both"/>-->
</profile>
</configuration>
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
