SIP-Transport Integration
You need Apache OpenMeetings version 2.1 to apply this guide!
You need Asterisk version 11 to apply this guide!
Here is instruction how-to set up red5sip transport integration with OpenMeetings on Ubuntu 12.10.
Prerequisites
Run the commands
sudo apt-get update && sudo apt-get upgrade sudo apt-get install build-essential linux-headers-`uname -r` libxml2-dev libncurses5-dev libsqlite3-dev sqlite3 openssl libssl-dev
ODBC Setup
Run the commands
sudo apt-get update sudo apt-get install unixODBC unixODBC-dev libmyodbc
Set up Asterisk connector:
Modify file /etc/odbc.ini as follows: (replace USER, PASSWORD and Socket with values relative to your system)
Modify file
/etc/odbcinst.ini as follows: (replace the path to the *.so files below with the real paths on your system)
(The path below is for x32 server, x64 version is most probably located at /usr/lib/x86_64-linux-gnu/odbc)
Run the following command to ensure everything works as expected:
Modify file /etc/odbc.ini as follows: (replace USER, PASSWORD and Socket with values relative to your system)
[asterisk-connector] Description = MySQL connection to 'openmeetings' database Driver = MySQL Database = openmeetings Server = localhost USER = root PASSWORD = Port = 3306 Socket = /var/run/mysqld/mysqld.sock
(The path below is for x32 server, x64 version is most probably located at /usr/lib/x86_64-linux-gnu/odbc)
[MySQL] Description = ODBC for MySQL Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so FileUsage = 1
echo "select 1" | isql -v asterisk-connector
Building and setting up Asterisk
Run the commands
sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.2.1.tar.gz sudo tar -xvzf asterisk-11.2.1.tar.gz cd ./asterisk-11.2.1 sudo make clean sudo ./configure sudo make sudo make install sudo make samples sudo make config sudo service asterisk start
Configure Asterisk
Enable asterisk ODBC module:
Modify "[modules]" section of /etc/asterisk/modules.conf as follows:
Add/uncomment the following lines
Modify "[modules]" section of /etc/asterisk/modules.conf as follows:
Add/uncomment the following lines
preload => res_odbc.so preload => res_config_odbc.so
Create/update "[asterisk]" section in
/etc/asterisk/res_odbc.conf:
[asterisk] enabled => yes dsn => asterisk-connector pre-connect => yes
Modify
/etc/asterisk/sip.conf
Add/uncomment the following line:
Increase maxexpiry value to 43200:
Add user for the "SIP Transport":
Add/uncomment the following line:
videosupport=yes rtcachefriends=yes
maxexpiry=43200
[red5sip_user] type=friend secret=12345 disallow=all allow=ulaw allow=h264 host=dynamic nat=force_rport,comedia context=rooms-red5sip
Add next lines into the
/etc/asterisk/extconfig.conf:
[settings] sippeers => odbc,asterisk,sipusers
Modify
/etc/asterisk/extensions.conf
Add the following section:
Add the following section:
[rooms] exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail) exten => _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})}) exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user) exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN}) exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,) exten => _400X!,n,Hangup exten => _400X!,n(notavail),Answer() exten => _400X!,n,Playback(invalid) exten => _400X!,n,Hangup [rooms-originate] exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user) exten => _400X!,n,Hangup [rooms-out] ; ***************************************************** ; Extensions for outgoing calls from Openmeetings room. ; ***************************************************** [rooms-red5sip] exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail) exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) exten => _400X!,n(notavail),Hangup
Modify
/etc/asterisk/confbridge.conf
Add/Modify the following secions:
Add/Modify the following secions:
[general] [red5sip_user] type=user marked=yes dsp_drop_silence=yes denoise=true [sip_user] type=user end_marked=yes wait_marked=yes music_on_hold_when_empty=yes dsp_drop_silence=yes denoise=true [default_bridge] type=bridge video_mode=follow_talker
To enable Asterisk Manager API modify
/etc/asterisk/manager.conf
Add/Modify the following sections:
Add/Modify the following sections:
[general] enabled = yes webenabled = no port = 5038 bindaddr = 127.0.0.1 [openmeetings] secret = 12345 deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 read = all write = all
Update Openmeetings with creadentials for Asterisk manager. Modify
/opt/red5/webapps/openmeetings/WEB-INF/openmeetings-applicationContext.xml
find <bean id="sipDao" class="org.apache.openmeetings.data.conference.dao.SipDao"> uncomment its parameters and set it to your custom values.
find <bean id="sipDao" class="org.apache.openmeetings.data.conference.dao.SipDao"> uncomment its parameters and set it to your custom values.
IMPORTANT: this step should be done BEFORE system install/restore otherwise all SIP related room information will be lost
Restart asterisk:
service asterisk restart
Setup red5sip transport
Download red5sip from
http://red5phone.googlecode.com/svn/branches/red5sip_2.1
Build with Apache Ant
ant
Insert proper values to the
/opt/red5sip/settings.properties
red5.host=127.0.0.1 # red5 server address om.context=openmeetings # Openmeetings context red5.codec=asao red5.codec.rate=22 # should correlate with mic settings in public/config.xml sip.obproxy=127.0.0.1 # asterisk adderss sip.phone=red5sip_user # sip phone number sip.authid=red5sip_user # sip auth id sip.secret=12345 # sip password sip.realm=asterisk # sip realm sip.proxy=127.0.0.1 # address of sip proxy rooms.forceStart=no # TBD rooms=1 # TBD (not in use)
Add red5sip to autostart:
sudo cp /opt/red5sip/red5sip /etc/init.d/ sudo chmod a+x /etc/init.d/red5sip sudo update-rc.d red5sip defaults
Start openmeetings
service red5 start
Start red5sip
service red5sip start