1. 檢查是否已經安裝kernel-devel<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />
# rpm -q kernel-devel
2. 檢查是否已經安裝以下的輔助軟件包
rpm -q bison
rpm -q bison-devel
rpm -q ncurses
rpm -q ncurses-devel
rpm -q zlib
rpm -q zlib-devel
rpm -q openssl
rpm -q openssl-devel
rpm -q gnutls-devel
rpm -q gcc
rpm -q gcc-c++
3. 下載libpri,zaptel,asterisk軟件包
http://ftp.digium.com/pub/libpri/releases/libpri-1.4.1.tar.gz
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.4.tar.gz
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.8.tar.gz
http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.2.tar.gz
http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz
將下載的軟件包放到自定義的目錄下; 例如:/usr/src/
4. 解壓縮軟件包
#cd /usr/src
# tar -xvzf libpri-<?xml:namespace prefix = st1 ns = "urn:schemas-microsoft-com:office:smarttags" />1.4.1.tar.gz
# tar -xvzf zaptel-1.4.4.tar.gz
# tar -xvzf asterisk-1.4.8.tar.gz
# tar -xvzf asterisk-addons-1.4.2.tar.gz
# tar -xvzf asterisk-sounds-1.2.1.tar.gz
# rm -r *.gz (刪除壓縮包,可以不執行這一步)
5.安裝libpri
# cd libpri-1.4.1
# make
# make install
6.安裝zaptel
# cd ..
# cd zaptel-1.4.4
# ./configure
# make (make is OK, make26 seems to be useless for recent zaptel versions)
# make install
# make config (to add zaptel startup script /etc/init.d/zaptel)
7.安裝asterisk
# cd ..
# cd asterisk-1.4.8
# ./configure
# make
# make install
# make samples
8. 安裝asterisk-addons
# cd ..
# cd asterisk-addons-1.4.2
# ./configure
# make
# make instal
l如果在make install的時候出現錯誤“cannot stat `.libs/libchan_h323.so.1.0.1'”解決辦法：
#cd /usr/src/asterisk-addons/asterisk-ooh323c/.libs/#ln -s libchan_h323.1.0.1 libchan_h323.so.1.0.1 (小寫的L)
# make samples
9.安裝asterisk-sounds
# cd ..
# cd asterisk-sounds-1.2.1
# make# make install
10.啓動Asterisk
asterisk -vvvvvvvvvvvvvvvcg 執行後會出現很多提示，注意如果裏面有紅色的ERROR字樣說明有問題。如果執行成功，最後會屏幕停在：Asterisk Ready*CLI>
11.註冊SIP用戶
步驟:
-Registering a user in sip.conf-Registering a number in extensions.conf
11.1 Go to the /etc/asterisk directory on your Asterisk server. Open sip.conf and register the following test user.
<?xml:namespace prefix = v ns = "urn:schemas-microsoft-com:vml" /> 
Note that ALL characters after ‘;’ (semi-colomn) are interpreted by Asterisk as a comment.
[ivan]
This means we are registering user 'ivan'
type=friend
This means the user can place or receive calls. For INBOUND calls only, use ‘peer’ as type. For outbound calls only use ‘user’ as type.
username=ivan
This declares that our user will be named 'ivan'
secret=pwd_ivan
This creates the password for the user to login/authenticate on Asterisk
host=dynamic
This sets dynamic IP for the host. You may also define this as a static IP
context=tutorial
This defines the dial context for the user which in this case is tutorial. In Asterisk, outgoing numbers are divided in groups called contexts in order to separate/define different needs for different user types. For example, a context for local calls, another for within the city, and another for international calls and so on.
This completes the user registration process on Asterisk. Now let us register an extension for the user.
11.2 Go to /etc/asterisk and open extensions.conf
[tutorial]
This registers the context ‘tutorial’
exten => 1234,1,Dial(SIP/ivan)
when dialing number 1234, Asterisk will first Dial the user xlite through SIP protocol. The command is : exten => number, priority, Dial(protocol/user). The priority determines the sequence in which the extensions will be executed.
We now have a registered SIP user and extension on Asterisk. Follow this same process to register another SIP user and extension in order to place test calls.
Start by registering the second user in the same way in /etc/asterisk/sip.conf
Register the extension(4321) in /etc/asterisk/extensions.conf in the same context = tutorial.
Now when user 'ivan' or any other user from the tutorial context dials 4321, the user 'test' will be called.
3. The final step is to register the user to a compatible softphone. However, softphones will be reviewed later. For now, just make sure you have registered the users and extensions.
More detailed configuration information for a series of phones can be found here:
01. asterisk with softphones http://www.asteriskguru.com/tutorials/configuration_asterisk_softphone.html
12.安裝、啓動X-Lite
# cd /opt/xten-xlite./xtensoftphone
少了個libstdc++.so.5,下載裝上就可以了；
13.錯誤處理
13.1 問題：軟電話登錄時出現“Login Timeout.Contact Firewall..”
解決：桌面|管理|安全級別和防火牆|其他端口|添加5060端口，重啓Asterisk和軟電話