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
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
see sample config for new options.
the dingaling library has changed so you must rebuild it
rm libs/libdingaling/.complete
make installall
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2251 d0543943-73ff-0310-b7d9-9358b9ac24b2
To Test:
uncomment or add from modules.conf
make installall again to compile it
uncomment the load line from freeswitch.xml
the default values are to bind to 127.0.0.1 port 8021
telnet to port 8021
enter "auth ClueCon" to authenticate
from here you can do the following:
*) events [xml|plain] <list of events to log or all for all>
*) noevents
*) log <level> // same as the console.conf values
*) nolog
*) api <command> <arg>
*) exit
there is a perl client in scripts/socket called fs.pl
with the module up and loaded:
cd scripts/socket
perl fs.pl <optional log level>
you can enter a few api commands like "show or status"
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2047 d0543943-73ff-0310-b7d9-9358b9ac24b2
DTMF:
1 = volume down
2 = volume default
3 = volume up
4 = gain down
5 = gain default
6 = gain up
7 = energy level down
8 = energy level default
9 = energy level up
* = toggle mute + deaf
0 = toggle mute
# = quit
new api commands to adj volume, gain and energy
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@1787 d0543943-73ff-0310-b7d9-9358b9ac24b2
