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標題: 我是用這個Compile Asterisk 1.8 的 [打印本頁]
作者: bubblestar    時間: 2011-1-4 18:27     標題: 我是用這個Compile Asterisk 1.8 的

我是用這個Compile Asterisk 1.8 的，希望大家有用。

根據各位不同的需要，可能不需要yum install 太多以下東西:
  1. yum -y install gcc gcc-c++ libxml2-devel ncurses-devel
  2. yum -y install openssl-devel mysql-devel sqlite-devel unixODBC-devel libtool-ltdl-devel
  3. yum -y install freetds-devel libvorbis-devel alsa-lib-devel bluez-libs-devel curl-devel libtiff-devel  libssl-dev
  4. yum -y install libusb-devel gmime-devel net-snmp-devel flex bison lua-devel subversion
  5. yum -y install make gnutls-devel gnutls-utils kernel-devel newt-devel mlocate lynx tar wget nmap
  6. yum -y install bzip2 mod_ssl crontabs vixie-cron libxml2 texinfo neon neon-devel


  9. # for Gtalk support
  10. cd /usr/src
  11. wget http://iksemel.googlecode.com/files/iksemel-1.4.tar.gz
  12. tar zxvf iksemel-1.4.tar.gz
  13. cd iksemel-1.4
  14. ./configure
  15. make && make check && make install
  16. ldconfig -p | grep semel
  17. echo "/usr/local/lib" >> /etc/ld.so.conf.d/iksemel.conf
  18. ldconfig
  19. ldconfig -p | grep semel
  20. cd ..

  22. # install dahdi
  23. wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
  24. tar zxvf dahdi-linux-complete-current.tar.gz
  25. cd dahdi-linux-complete-2.4.0+2.4.0
  26. make && make install && make config
  27. chkconfig dahdi on
  28. service dahdi start
  29. cd ..

  31. #If you are using E1 cards you need to install LIBPRI.
  32. wget http://downloads.asterisk.org/pub/telephony/libss7/libss7-1.0.2.tar.gz
  33. tar zxvf libss7-1.0.2.tar.gz
  34. cd libss7-1.0.2
  35. make && make install
  36. cd ..

  38. wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.12-beta3.tar.gz
  39. tar zxvf libpri-1.4.12-beta3.tar.gz
  40. cd libpri-1.4.12-beta3
  41. make && make install
  42. cd ..

  44. rpm -ivh http://ftp.yz.yamagata-u.ac.jp/pub/linux/fedora/epel/5/i386/libical-0.43-4.el5.i386.rpm
  45. rpm -ivh http://ftp.yz.yamagata-u.ac.jp/pub/linux/fedora/epel/5/i386/libical-devel-0.43-4.el5.i386.rpm

  47. cd ..

  49. #optional; If you also want to build the PDF manual (but you don't need to, you will need an optional tool called rubber that compiles Latex into a PDF
  50. wget http://launchpad.net/rubber/trunk/1.1/+download/rubber-1.1.tar.gz
  51. tar zxvf rubber-1.1.tar.gz
  52. cd rubber-1.1
  53. ./configure && make && make install
  54. cd ..

  56. #install asterisk
  57. wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.1.1.tar.gz
  58. tar zxvf asterisk-1.8.1.1.tar.gz
  59. cd asterisk-1.8.1.1
  60. ./configure
  61. #this is only for format MP3 - SVN required
  62. contrib/scripts/get_mp3_source.sh
  63. contrib/scripts/get_ilbc_source.sh
  64. make menuselect
  65. make
  66. make install
  67. make samples
  68. make progdocs
  69. make config
  70. make install-logrotate

  72. chkconfig asterisk on
  73. service dahdi start
  74. service asterisk start
複製代碼
作者: ckleea    時間: 2011-1-4 21:12

回復 1# bubblestar

What is your status of Calendar? Are you able to load the google calendar?
作者: bubblestar    時間: 2011-1-4 21:26

Not yet !  Still studying the setup instructions.
作者: ckleea    時間: 2011-1-4 22:12

If you properly  set up, in CLI, calendar show calendars will show up proper entry
作者: bubblestar    時間: 2011-1-5 15:06

本帖最後由 bubblestar 於 2011-1-5 15:20 編輯

成功利用Google Calendar 在 Asterisk 1.8 之下使用 Wake up Call.  

先在網上的Google Calendar 隨便設定一個約會或什麼東西也可以。記著要剔取Alarm 或 Reminder，當時間一到，Asterisk 1.8 Server 即刻會致電給你作為提示，其中一個用途可當作Wake up call，貪睡的朋友或在公司蛇王的朋友，到時到候它會提你起身啦

/etc/asterisk/calendar.conf

[myGoogleCal]
type = caldav
url = https://www.google.com/calendar/ ... e@gmail.com/events/
user = your_username@gmail.com
secret = your_top_secret_password
refresh = 15
timeframe = 60
autoreminder = 10
channel = SIP/6688
waittime = 30
app = Playback
appdata = this-is-yr-wakeup-call       ; 這裡如果選擇播放音樂檔應該沒有問題。
作者: ckleea    時間: 2011-1-5 22:19

One naive question. I am unable to use SIP/exten. Instead I can use Local/exten. Do I need to register every extension in the sip.conf. A
作者: bubblestar    時間: 2011-1-5 22:42

In my case, SIP/exten is one of the real registered extension in the sip.conf.  

I have not tried Local/exten, so no ideas about that.  Is "Local" one of your context in extension.conf or sip.conf?  
I think the point is your designated extension must be reachable and can get the incoming call through ringing.
作者: bubblestar    時間: 2011-1-5 22:47

用了此功能，偷懶的朋友也可以變成看似忙碌的員工了。只要在Google Calendar 設定好時間，咁樣便經常會有電話找大家。 真亦假時，假亦真，老板們真係多得呢個功能唔少啦!
作者: bubblestar    時間: 2011-1-7 12:18

回復 6# ckleea


    It seems Local/Channel is a new feature introduced in Asterisk 1.8.  

    http://ofps.oreilly.com/titles/9780596517342/ch10.html#id3174771
作者: ckleea    時間: 2011-1-7 13:53

回復 9# bubblestar


    I have no problem but from time to time, if I use Dial (SIP/{exten}), sometimes it does not work
作者: bubblestar    時間: 2011-1-7 15:18

本帖最後由 bubblestar 於 2011-1-7 15:19 編輯
回復  bubblestar


    I have no problem but from time to time, if I use Dial (SIP/{exten}), someti ...
ckleea 發表於 2011-1-7 13:53



    Double check if you have added $ in front of {EXTEN}, i.e., SIP/${EXTEN}
作者: ckleea    時間: 2011-1-9 05:57

回復 7# bubblestar


    So, do you set up your extension under GUI or just paste the codes into sip.conf? I believe I can't use SIP/6000 is due to the setup under GUI. The information is in users.conf not sip.conf.

Then what is the purpose of GUI?
作者: bubblestar    時間: 2011-1-9 10:32

I, myself, do not use SIP/6000.  As mentioned before, extension 6000 is predefined to some other purpose, probably for auto-attendant.

I think we can adjust or fix to use extension 6000 again.
作者: ckleea    時間: 2011-1-9 17:54

回復 13# bubblestar

My 6000 is assigned to the Siemens IP phone. Works without problem.
Only I did not put any extension setting in SIP.conf.

My point is since we need to add the extensions setting in sip.conf and iax.conf. What is the point of using GUI to create users/extensions? Only you can see them under GUI.
作者: bubblestar    時間: 2011-1-9 21:53

Maybe I get you wrong.

OK.  As far as I know,if we use GUI to create extensions, it is NOT necessary to put those extensions in SIP.conf since they will be included in USERS.conf.  As I am not using GUI for Asterisk 1.8, all my extensions are put inside SIP.conf and do not need to touch users.conf.

Basically, we should handle extensions for IAX in the same fashion.  However, I notice from the Internet that we might face some strange problems if we check both IAX and SIP in GUI.  Hence, it is suggested that we'd better create IAX and SIP extensions separately to avoid confusion.

Those information indicates that GUI (at least Asterisk's own version) is really not the best choice at this moment.  Elastix, FreePBX, Trixbox and PiaF outperform in this respect.
作者: ckleea    時間: 2011-1-10 23:06

I have just migrated my original asterisk to the new D525 ATOM. Just few recompilations of dahdi modules as I am now default to use xen kernel



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