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fixed bug in AST_DB_optimize script

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fixed issues with tz_divide script
docs updates

git-svn-id: svn://192.168.202.10@708 3d104415-ff17-0410-8863-d5cf3c621b8a
This commit is contained in:
mattf
2007-11-09 22:35:58 +00:00
parent bb77c6a404
commit 5a7674b47b
5 changed files with 122 additions and 81 deletions
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docs/SCRATCH_INSTALL.txt
+36 -27
View File
@@ -841,15 +841,15 @@ with the following patch:
- mkdir /usr/src/asterisk
- cd /usr/src/asterisk
A. if you want 1.2 release (reliable with new features):
- wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
- wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
- wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz
- gunzip asterisk-1.2.17.tar.gz
- tar xvf asterisk-1.2.17.tar
- gunzip zaptel-1.2.16.tar.gz
- tar xvf zaptel-1.2.16.tar
- gunzip libpri-1.2.4.tar.gz
- tar xvf libpri-1.2.4.tar
- wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.24.tar.gz
- wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.21.tar.gz
- wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.5.tar.gz
- gunzip asterisk-1.2.24.tar.gz
- tar xvf asterisk-1.2.24.tar
- gunzip zaptel-1.2.21.tar.gz
- tar xvf zaptel-1.2.21.tar
- gunzip libpri-1.2.5.tar.gz
- tar xvf libpri-1.2.5.tar
B. if you want latest SVN_1.2 version (release tree with new patches)
- svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
- svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
@@ -872,15 +872,15 @@ with the following patch:
edit this line at the top and just add a hash # in front of it as shown:
#PBX_LIBS+=$(shell $(CROSS_COMPILE_BIN)gtk-config --cflags >/dev/null 2>/dev/null && echo "pbx_gtkconsole.so")
- cd ./zaptel-1.2.16
- cd ./zaptel-1.2.21
- make clean
- make
- make install
- cd ../libpri-1.2.4
- cd ../libpri-1.2.5
- make clean
- make
- make install
- cd ../asterisk-1.2.17
- cd ../asterisk-1.2.24
- (1.2 tree) If you want to include Answering Machine Detection ability
you will need to download app_amd.c and amd.conf and alter the
apps/Makefile to compile it properly
@@ -895,7 +895,7 @@ with the following patch:
- wget http://www.eflo.net/files/amd2.conf
- mkdir /etc/asterisk
- mv amd2.conf /etc/asterisk/amd.conf
*OPTIONAL*(1.2.12.1 thru 1.2.17) apply the meetme volume control patch
*OPTIONAL*(1.2.12.1 thru 1.2.24) apply the meetme volume control patch
*Different patches available for 1.2.7.1 through 1.2.14
- wget http://www.eflo.net/files/meetme_volume_control_1.2.16.patch
- patch -p1 < ./meetme_volume_control_1.2.16.patch
@@ -907,7 +907,7 @@ with the following patch:
- patch -p1 < ./cli_chan_concise_delimiter.patch
- File to patch: cli.c
*OPTIONAL*(1.2.14 thru 1.2.18) rewrite of waitforsilence
*OPTIONAL*(1.2.14 thru 1.2.24) rewrite of waitforsilence
- wget http://www.eflo.net/files/app_waitforsilence.c
- mv app_waitforsilence.c apps/app_waitforsilence.c
@@ -923,17 +923,6 @@ with the following patch:
- make install
- make samples # this makes sample conf files (only use for new installs)
- (1.2 tree) EXPERIMENTAL AND NOT RECOMMENDED!!! If you want to use
app_conference instead of meetme for VICIDIAL then follow these
instructions:
- cd /usr/src/asterisk
- wget http://www.eflo.net/files/VD_app_conference_0.6.zip
- unzip VD_app_conference_0.6.zip
- cd app_conference
- make clean
- make
- make install
- modprobe zaptel # this loads the zaptel module
- install the module for the digium device that you are using, we are
using the T100P single span T1 card so we use:
@@ -961,7 +950,7 @@ them:
- If you have chosen a Sangoma T1/E1 or analog card, you will need to
follow their instructions for installation of their driver software
LATEST Sangoma Wanpipe drivers:
ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-2.3.4-13.tgz
ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.2.1.tgz
- now your asterisk installation is built and loaded and it's time to
configure it.
@@ -1000,6 +989,26 @@ NOTE: you can get an X100P through ebay for $10-$30 that will work with Asterisk
modprobe zaptel
modprobe wcfxo
If you will be doing native music-on-hold for your inbound calls, you will need
musiconhold audio files to be converted to native formats like GSM, ULAW and ALAW:
cd /var/lib/asterisk/mohmp3/
mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-sunshine.mp3 > /var/lib/asterisk/mohmp3/fpm-sunshine.raw
sox -r 44100 -w -s -c 1 fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav
sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm
sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm
mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-calm-river.mp3 > /var/lib/asterisk/mohmp3/fpm-calm-river.raw
sox -r 44100 -w -s -c 1 fpm-calm-river.raw -r 8000 -c 1 fpm-calm-river.wav
sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm
sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm
mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-world-mix.mp3 > /var/lib/asterisk/mohmp3/fpm-world-mix.raw
sox -r 44100 -w -s -c 1 fpm-world-mix.raw -r 8000 -c 1 fpm-world-mix.wav
sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm
sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm
rm -f *.raw
mkdir ../orig-mp3
mv -f *.mp3 ../orig-mp3/
PHASE 5: CONFIGURING ASTERISK AND YOUR SIP PHONES
@@ -1660,7 +1669,7 @@ UPDATE mysql.user set password=OLD_PASSWORD('1234') where user='cron';
use asterisk;
\. /usr/src/astguiclient/MySQL_AST_CREATE_tables.sql
\. /usr/share/astguiclient/MySQL_AST_CREATE_tables.sql
or you may need to run this if you get an error:
\. /usr/src/astguiclient/trunk/extras/MySQL_AST_CREATE_tables.sql