Asterisk-Software

• vergl. http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM

• Siehe: http://www.das-asterisk-buch.de/beta-test/installation-asterisk-1.2.html
• apt-get install openssl openssl-devel zlib zlib-devel krb5-devel ncurses-devel
• cd /usr/src
• wget http://ftp.digium.com/pub/asterisk/asterisk-1.2-current.tar.gz
  • oder lokale Kopie: asterisk-1.2.24.tar.gz * cd asterisk-1.2.14
• make
  • Läuft durch
• make install
• make samples

• ast_prompts_de_v2_0.tar.gz von http://www.beronet.net/
  • lokale Kopie: ast_prompts_de_v2_0.tar.gz
• tar xvzf ast_prompts_de_v2_0.tar.gz
• cd ast_prompts_de_v2_0/var/lib/asterisk/sounds
• cp -av * /var/lib/asterisk/sounds
• cd /var/lib/asterisk/sounds
• chmod -R 644 de/ digits/de/ phonetic/de/ letters/de/
• vi /etc/astersisk/sip.conf
  • sip.conf:language=de
• vi /etc/asterisk/extensions.conf
  • exten ⇒ 2200,1,Set(LANGUAGE()=de)
• vi /etc/asterisk/indications.conf

  [general]
  country=de

Ohne deutsche Sprachdateien Probleme beim Mailbox-System. Es wurde versucht Datei „digits/1F.ulaw“ bzw „1F.gsm“ abzuspielen, die es aber nicht gibt (im Standard).

• cd /usr/src
• wget http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz
• tar xvzf asterisk-1.4-current.tar.gz
• cd asterisk-1.4.<TAB>
• ./configure
• make
• make install

Deutsche Sprachdateien: wie gehabt.

• lösche „Sample entries“ in /etc/asterisk/extensions.conf

für Fedora:

#!/bin/bash */
/*  */
/*  asterisk      Startup script for the asterisk server */
/*                Klaus.Franken@StrukturPunkt.de */
/*  */
/*  chkconfig: - 85 15 */
/*  description: Asterisk PBX */
/*  processname: asterisk */
/*  config: /etc/asterisk/sip.conf */
/*  config: /etc/asterisk/extensions.conf */
/*  pidfile: /var/run/asterisk.pid */

/*  Source function library. */
. /etc/rc.d/init.d/functions

if [-f /etc/sysconfig/asterisk ]; then
        . /etc/sysconfig/asterisk
fi

prog=asterisk
pidfile=${PIDFILE-/var/run/asterisk.pid}
pidfile2=${PIDFILE-/var/run/asterisk_loop.pid}
RETVAL=0

start() {
        echo -n $"Starting $prog: "
        modprobe ztdummy # for conferences
        daemon $prog
        RETVAL=$?
        if test -r /var/www/webtel/kibeon/cgi-bin/asterix_loop.sh; then
            sleep 1
            /var/www/webtel/kibeon/cgi-bin/asterix_loop.sh >> /var/log/asterisk_loop.log 2>&1 &
            loopPid=$!
            echo $! > $pidfile2
        fi
        echo
        #[[:intern|$RETVAL = 0 ]] && touch ${lockfile}
        asterisk -rx "core set verbose 9"
        return $RETVAL
}
stop() {
        echo -n $"Stopping Asterisk"
        asterisk -rx "stop now"
        loopPid=''
        test -r $pidfile2 && loopPid=`cat $pidfile2`
        test ! -z $loopPid && kill $loopPid
        rm -f $pidfile2
        echo
}
reload() {
        echo -n $"Reloading Asterisk-Configuration"
        echo
        asterisk -rx "reload"
        echo
}
graceful() {
        echo -n $"Restarting  Asterisk gracefully"
        asterisk -rx "restart gracefully"
        echo
        channels
}
channels() {
        echo $"activ channels:"
        asterisk -rx "sip show channels"
        echo
}

/*  See how we were called. */
case "$1" in
  start)
        start
        ;;
  stop)
        stop
        ;;
  status)
        channels
        status $prog
        RETVAL=$?
        ;;
  restart)
        stop
        sleep 1
        start
        ;;
  condrestart)
        if [-f ${pidfile} ] ; then
                stop
                sleep 1
                start
        fi
        ;;
  reload)
        reload
        ;;
  channels)
        channels
        ;;
  graceful|gracefully)
        graceful
        ;;
    * )
        echo $"Usage: $prog {start|stop|restart|condrestart|reload|status|channels|graceful}"
        exit 1
esac

exit $RETVAL

• vi /etc/firehol/firehol.conf

server_rtpproxy_ports="udp/35000:35010"
client_rtpproxy_ports="default"
...
interface any world
...
        server sip accept
        server rtpproxy accept

• vi /etc/asterisk/rtp.conf

rtpstart=35000
rtpend=35010

• muss zur Firewall passen