【角色茶桌】——Asterisk 1.8 + Siptosis彻底安装成功!
本帖最後由 角色 於 2011-10-30 07:58 編輯
Installation Reference
Please make use of the following reference link in the course of installing your Skype for Asterisk using Siptosis gateway
http://www.mhspot.com/sts/sts_install_centos.html
CentOS 5.X
Skype
要在坊间搜索skype_static-2.1.0.81.tar.bz2的文件,在Skype的网站的Skype for Linux是给CentOS 6.0用的。
Siptosis (Single Skype Account)
http://www.mhspot.com/sts/siptosis_download.php
Java for Linux
The JRE java rpm is obtainable in the following link:
http://www.java.com/en/download/manual.jsp
The basic working principle of Skype for Asterisk:
There is a Skype client on a Linux box. In order to interface with other application, a Java for Linux is employed. Siptosis is able to talk to Linux-based Skype via Java.
Installation procedures
1. Down the file jxvf skype_static-2.1.0.81.tar.bz2 to /usr/src/skype directory
2. Unzip and untar the file skype_static-2.1.0.81.tar.bz2- tar jxvf skype_static-2.1.0.81.tar.bz2
複製代碼 to /usr/src/skype directory
3. Create symbolic links- ln -s /usr/src/skype /usr/share/skype
- ln -s /usr/src/skeype/skype /usr/bin/skype
複製代碼 4. Install X Window and other packages- yum install libXv
- yum install libXScrnSaver
- yum groupinstall "X Window System"
- yum install alsa-lib
複製代碼 4. If you are using twm X Window Manager, you have to make the following adjustment such that you would not need to place the popup window manually. Login root before carrying out the following changes:- vi /etc/X11/twm/system.twmrc - add RandomPlacement above NoGrabServer line.
- cp /etc/X11/twm/system.twmrc /root/.twmrc
複製代碼 如果安装好以后,下面的files (/usr/src/siptosis folder)经常会看:
5. Test the X Window and Skype- su -l root
- [root-password]
- cd /usr/src
- // To start X Window
- startx
- // To start Skype
- skype
複製代碼 6. Installation of Java for Linux
7. Installation of Siptosis in /usr/src/siptosis
After installation, please read the file /usr/src/siptosis/readme.txt which gives you more information for the installation.
8. Change the directory to /usr/src/siptosis and modify the following in ./siptosis.cfg- #Sample AUTO config with NO registration
- # username and password not important in this mode
- # Set to available port to transport SIP messages on siptosis computer
- host_port=5070 // Modify this port number if neccessary
- username=skypests // Modify this if neccessary
- passwd=unimportantpassword
- do_register=no
- # --- end of NO registration example ---
複製代碼 9. Edit the Asterisk server which communicate with the Siptosis. Please note the Asterisk server and Siptossis may not the same location (i.e. same IP)- [skypests]
- username=skypests
- type=friend
- secret=skype
- host=192.168.1.103
- nat=no
- dtmfmode=auto
- ;canreinvite=yes (use only if you understand what it does - does not work well with ilbc and speex codecs)
- canreinvite=no
- ;port should not be needed if you register with the PBX - some have said it's needed??
- ;port=siptosishostport
- port=5070
- qualify=yes
- defaultip=192.168.1.103
- incominglimit=1
- outgoinglimit=1
- call-limit=1
- busylevel=1
- context=from-skype
複製代碼 Sip
SipToSkypeAuth.props
to forward and authorize SIP calls to desired Skype destinations. (Most users will only use: *,*,localnet,calleeid)
SipOutDialingRules.props
Skype
SkypeOutDialingRules.props
for any Skype dialing rules/transforms wanted. When you connect to the Skype, press 55 for calling echo123, press 56 for calling skype_name_1.- #you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
- ^55$:echo123
- ^56$:skype_name_1
- ^57$:skype_name_2
- #send callme im to echo123 to get a call back from the test service
- ^559$:im:echo123:callme
複製代碼 SkypeToSipAuth.props
to forward skype calls to SIP destinations (failure to do this step will cause all incoming Skype calls to get the invalid destination. The first two line show the default incoming Skype calling, which goto sip:1911@192.168.1.5:5060. The last two line is never executed, which is given for reference only.- #*,sip:3000@192.168.1.103:5228
- *,sip:1911@192.168.1.5:5060
- #Default: all incoming skype callers get the invalid destination message
- *,play:clips/invalidDest.wav
複製代碼 角色 |