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標題: Compiling Asterisk 1.8 on Centos 5.5 32/64 bits [打印本頁]

作者: ckleea 時間: 2010-10-30 07:11 標題: Compiling Asterisk 1.8 on Centos 5.5 32/64 bits

本帖最後由 ckleea 於 2010-10-30 16:46 編輯

Just come across
http://astrecipes.net/index.php?n=398

Compile Asterisk 1.8 with iCal, GTalk, SNMP, MySQL, cURL and DAHD

This site describes how to compile in 64 bits linux.

I have modified to 32 bits. Gtalk can now work.
I

yum install gcc-c++ make gnutls-devel kernel-devel
yum install libxml2-devel ncurses-devel subversion doxygen texinfo
yum install curl-devel net-snmp-devel neon-devel
cd /usr/src
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.4.0+2.4.0.tar.gz
tar zxvf dahdi-linux-complete-2.4.0+2.4.0.tar.gz
cd dahdi-linux-complete-2.4.0+2.4.0
make
make install
[color=Red]make config[/color] ; missing from site.
cd ..
http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.12-beta2.tar.gz
tar zxvf libpri-1.4.12-beta2.tar.gz
cd libpri-1.4.12-beta2
make
make install
cd ..
rpm -Uvh http://dl.atrpms.net/all/libiksemel3-1.4-2_2.el5.i386.rpm
wget http://sourceforge.net/projects/freeassociation/files/libical/libical-0.44/libical-0.44.tar.gz/download
tar zxvf libical-0.44.tar.gz
cd libical-0.44
./configure --enable-shared
make
make install
cd ..
wget http://launchpad.net/rubber/trunk/1.1/+download/rubber-1.1.tar.gz
tar zxvf rubber-1.1.tar.gz
cd rubber-1.1
./configure
make
make install
cd ..
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.0.tar.gz
tar zxvf asterisk-1.8.0.tar.gz
cd asterisk-1.8.0
./configure
make menuselect
make
#this is ony for format MP3 - SVN required
contrib/scripts/get_mp3_source.sh
make install
make samples
make progdocs

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作者: 角色 時間: 2010-10-30 10:18

How was the result? Are you able to make it work?

YH

作者: ckleea 時間: 2010-10-30 10:20

Apparently works. I just install in VM. Need to add more modules before real test. But looks better when is in CLI. c.f. rpm methods

作者: ckleea 時間: 2010-10-30 10:47

本帖最後由 ckleea 於 2010-10-30 11:45 編輯

Still have errors in the GUI and DADHI
DADHI errors fixed. Forgot to make config after make install
Same for asterisk

作者: ckleea 時間: 2010-10-30 12:45

Updated. It works in APL. No GUI at the moment. I duplicate my IP01 setting into asterisk 1.8.

作者: bubblestar 時間: 2010-10-30 13:17

本帖最後由 bubblestar 於 2010-10-30 13:18 編輯

Speaking of installation on 64 bits system, it seems we may encounter problem for getting codec g723 and g729 as there are no ready-made codec for ATOM computer. I had tried to get them for my ATOM-Based Asterisk 1.4 or 1.6 but failed. I finally turned back to 32 bits for the time being.

Do you know whether these codecs are already included in Asterisk 1.8 or not ??

作者: ckleea 時間: 2010-10-30 13:37

本帖最後由 ckleea 於 2010-10-30 19:24 編輯

Not yet, just start to make it works. I only install 32 bits.

See my first post, keep updating.

Now Gtalk work.

ast18*CLI> module load chan_gtalk.so
Loaded chan_gtalk.so
== Parsing '/etc/asterisk/gtalk.conf': == Found
== Registered RTP glue 'Gtalk'
== Registered channel type 'Gtalk' (Gtalk Channel Driver)
Loaded chan_gtalk.so => (Gtalk Channel Driver)
JABBER: asterisk INCOMING:
JABBER: asterisk INCOMING:

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作者: ckleea 時間: 2010-11-3 21:55

本帖最後由 ckleea 於 2010-11-3 22:00 編輯

Another update, as brave as bubblestar, I upgrade my asterisk 1.6.2.12 to a compiled version of asterisk 1.8. The GUI works as well.

Everything seems ok though some errors seen in the log. I put in g723 and g729 codec as well

圖片附件: screenshot.9.jpg (2010-11-3 21:59, 213.48 KB) / 下載次數 1251
http://telecom-cafe.com/forum/attachment.php?aid=244&k=1f4702f3beb45998741e923e1bc93c55&t=1732347885&sid=7333d9

圖片附件: screenshot.8.jpg (2010-11-3 22:00, 74.43 KB) / 下載次數 1276
http://telecom-cafe.com/forum/attachment.php?aid=247&k=00b8c8df89ca744d61d04d45d13ead1d&t=1732347885&sid=7333d9

作者: ckleea 時間: 2010-11-3 22:53

Google voice works.

作者: bubblestar 時間: 2010-11-3 23:02

Google Voice can be used and dialed directly from hardphone? Congratulations! you did it.

作者: ckleea 時間: 2010-11-3 23:04

回復 10# bubblestar

Yes, it calls and when you are in voice.google.com, you can see the dial history.

作者: ckleea 時間: 2010-11-4 11:41

回復 6# bubblestar

I use 32 bits for my ATOM server. G729 and g&23 codecs loaded as before. But not yet tested.

作者: ckleea 時間: 2010-11-4 13:35

Manual of asterisk 1.8 attached for those interested.

附件: [Asterisk 1.8 Manual] asterisk.pdf (2010-11-4 13:35, 899.16 KB) / 下載次數 2919
http://telecom-cafe.com/forum/attachment.php?aid=249&k=820a27d16dd3c561386e207d3c2ee75d&t=1732347885&sid=7333d9

作者: lttliang 時間: 2010-11-5 18:00

Another update, as brave as bubblestar, I upgrade my asterisk 1.6.2.12 to a compiled version of ast ...
ckleea 發表於 2010-11-3 21:55

ip01里的code可否直接移到atom的asterisk用?

作者: bubblestar 時間: 2010-11-5 18:05

據 ckleea 兄所講，是可以的。我仲未試過，他有經驗，稍後我都會嘗試一下，可以的話就一定非常方便，無需再為每一個系統作相同的設定。

作者: ckleea 時間: 2010-11-5 20:26

回復 15# bubblestar

My asterisk 1.8 is just using the old 1.6 config files. Overwrite the default settings. So far it is ok but not for IAX. Something need to set.

作者: 電腦超人 時間: 2010-11-5 20:36

我去到make install不成功...

CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/usr/src/asterisk-1.8.0/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-1.8.0/menuselect'
[LD] aelparse.o aelbison.o pbx_ael.o hashtab.o lock.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o pval.o extconf.o -> aelparse
lock.o: In function `__ast_rwlock_trywrlock':
/usr/src/asterisk-1.8.0/utils/lock.c:1339: undefined reference to `ast_bt_get_addresses'
lock.o: In function `__ast_rwlock_tryrdlock':
/usr/src/asterisk-1.8.0/utils/lock.c:1273: undefined reference to `ast_bt_get_addresses'
lock.o: In function `__ast_rwlock_timedwrlock':
/usr/src/asterisk-1.8.0/utils/lock.c:1173: undefined reference to `ast_bt_get_addresses'
lock.o: In function `__ast_rwlock_timedrdlock':
/usr/src/asterisk-1.8.0/utils/lock.c:1073: undefined reference to `ast_bt_get_addresses'
lock.o: In function `__ast_rwlock_wrlock':
/usr/src/asterisk-1.8.0/utils/lock.c:959: undefined reference to `ast_bt_get_addresses'
lock.o:/usr/src/asterisk-1.8.0/utils/lock.c:845: more undefined references to `ast_bt_get_addresses' follow
collect2: ld returned 1 exit status
make[1]: *** [aelparse] Error 1
make: *** [utils] Error 2

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不知是甚麼問題...重覆試過數次也不行...
還在試...

作者: ckleea 時間: 2010-11-5 22:30

deselect utilities at the make menuconfig
You may have missing some libraries

作者: 電腦超人 時間: 2010-11-7 16:30

Haven't select Compiler Flags&Module Embedding...
And complete install~

But DADHI also cannot install...

GUI also have problems...

作者: 角色 時間: 2010-11-7 18:36

Google voice works.
ckleea 發表於 2010-11-3 22:53

How can you do it?

YH

作者: bubblestar 時間: 2010-11-7 18:47

回復 20# 角色

Asterisk 1.8 好似已經內置及提供此功能了，還有GTALK呢! 所以大家現在都想先睹及先試用為快。

呢度都有提到 Asterisk 1.8 and Google Voice

角色兄，你好嘛! 要多D來開咪講下野先至得啦，沒有你在的日子，少咗好多熱烈討論氣氛。

作者: 角色 時間: 2010-11-7 19:00

因为最近非常忙（主要都是写program工作），所以白天不能上来，不过，看到Asterisk 1.8可以做GV，那么我可能计划，买一台PC（Gigabyte D525 + Case + Hard disk 40G + Analogue cards （2 FXS， 2FX0）。

角色

作者: 角色 時間: 2010-11-7 19:05

我在想，在GV怎样才可以disable caller ID呢？

角色

作者: ckleea 時間: 2010-11-7 19:15

本帖最後由 ckleea 於 2010-11-7 21:09 編輯

回復 20# 角色

Need to enable gtalk and jabber using your gmail account. I understand google apps account does not work.
After gtalk and jabber works, add a simple dial plan for google voice.

e.g.

[CallingRule_gv]
exten => _813XXXXXXXXXX,1,Dial(gtalk/asterisk/+1${EXTEN:3}@voice.google.com)

作者: ckleea 時間: 2010-11-7 22:27

回復 16# ckleea

I found the problem of IAX is due to Zoiper client. It has two new versions recently. Both have problems in IAX connection. My older file works very well after reinstallation.

作者: lttliang 時間: 2010-11-12 15:59

因为最近非常忙（主要都是写program工作），所以白天不能上来，不过，看到Asterisk 1.8可以做GV，那么我可 ...
角色發表於 2010-11-7 19:00

我配左一部PC 用的是技嘉N230底板本来打算是用来做路由器用的，但是因为装左海蜘蛛软路由之后，效果并不理想，虽然可以将两条ADSL的速度提升双倍，但是只是在多线程软件中起到作用，用QQ或IE上传或download野时，都系单条ADSL的速度

所以我就用来装asterisk用了

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