Updated install documentation
Changed default server version to asterisk_version='1.4.21.2' git-svn-id: svn://192.168.202.10@1171 3d104415-ff17-0410-8863-d5cf3c621b8a
This commit is contained in:
mattf
+48
-82
@@ -1,4 +1,4 @@
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Asterisk/astguiclient install from existing server. v.2.2.0 2009-04-03
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Asterisk/astguiclient install from existing server. v.2.2.0 2009-06-28
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By the VICIDIAL group info@vicidial.com
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@@ -6,8 +6,16 @@ This document is meant to be more of an overview of a VICIDIAL/astGUIclient
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installation. If you are looking for more detail or are having problems with
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installation please read the SCRATCH_INSTALL.txt document.
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End-user Manuals for Agents and Managers are available from http://www.eflo.net
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There is also an easier Ubuntu install doc: Ubuntu_Install.txt
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If you want to skip the experience of installing step-by-step completely, take
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a look at our ISO installer: http://www.vicibox.com
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REQUIREMENTS:
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- root access on the server you want to install asGUIclient/VICIDIAL on to.
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- Zap trunks(T1/E1 or PSTN lines), IAX2 trunks or SIP trunks required
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@@ -18,10 +26,10 @@ REQUIREMENTS:
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- GNU screen 3.9.15 or greater must be installed
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- ntp must be installed and syncd to same time source as any other servers
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- Perl5 with several CPAN modules(Time::HiRes, DBI, DBD::mysql, Net::Telnet, Net::Server etc)
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- asterisk-perl 0.08, do NOT use the 0.09 version, it does not work with VICIDIAL
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- MySQL server or mysqlclientlibs, must be version 4.0.X or higher
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- asterisk-perl 0.08, do NOT use newer versions, they do not work with VICIDIAL
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- MySQL server or mysqlclientlibs, must be version 5.0.X or higher
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- If you will be installing Asterisk 1.4 releases you need Gnu Make 3.81
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- Apache/PHP server with php.ini memory_limit set to 48M
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- Apache/PHP server with php.ini memory_limit set to 48M or higher
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@@ -29,86 +37,42 @@ REQUIREMENTS:
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PHASE 1: INSTALLING ASTERISK
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1. follow these command line steps:
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mkdir /usr/src/asterisk
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cd /usr/src/asterisk
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wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.2.30.2.tar.gz
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wget http://downloads.digium.com/pub/zaptel/releases/zaptel-1.2.27.tar.gz
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wget http://downloads.digium.com/pub/libpri/releases/libpri-1.2.8.tar.gz
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gunzip asterisk-1.2.30.2.tar.gz
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tar xvf asterisk-1.2.30.2.tar
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gunzip zaptel-1.2.27.tar.gz
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tar xvf zaptel-1.2.27.tar
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gunzip libpri-1.2.8.tar.gz
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tar xvf libpri-1.2.8.tar
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cd ./zaptel-1.2.27
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wget http://download.vicidial.com/packages/newt-0.51.6.tar.gz
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gunzip newt-0.51.6.tar.gz
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tar xvf newt-0.51.6.tar
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cd newt-0.51.6
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./configure
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make
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make install
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cd ../
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ln -s /usr/lib/libnewt.so.0.51.6 /usr/lib/libnewt.so.0.51
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make clean
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make zttool
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make
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make install
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cd ../libpri-1.2.8
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make clean
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make
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make install
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cd ../asterisk-1.2.30.2
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(1.2 tree) If you want to include Answering Machine Detection ability
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you will need to download app_amd.c and amd.conf and alter the
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apps/Makefile to compile it properly
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cd apps
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wget http://www.eflo.net/files/app_amd2.c
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mv app_amd2.c app_amd.c
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vi Makefile
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replace this line(line 32):
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app_mixmonitor.so app_stack.so
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with this line:
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app_mixmonitor.so app_stack.so app_amd.so
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wget http://www.eflo.net/files/amd2.conf
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mkdir /etc/asterisk
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mv amd2.conf /etc/asterisk/amd.conf
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*OPTIONAL*(1.2.23 thru 1.2.30.2) apply the meetme DTMF passthru patch
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wget http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch
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patch -p1 < ./meetme_DTMF_passthru-1.2.23.patch
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- File to patch: app_meetme.c
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*OPTIONAL*(1.2.12.1 thru 1.2.30.2) apply the meetme volume control patch
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*Different patches available for 1.2.7.1 through 1.2.14
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wget http://www.eflo.net/files/meetme_volume_control_1.2.16.patch
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patch -p1 < ./meetme_volume_control_1.2.16.patch
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- File to patch: app_meetme.c
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cd ../
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(1.2 tree) apply the cli delimiter patch
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wget http://www.eflo.net/files/cli_chan_concise_delimiter.patch
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patch -p1 < ./cli_chan_concise_delimiter.patch
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- File to patch: cli.c
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*OPTIONAL*(1.2.14 thru 1.2.30.2) rewrite of waitforsilence
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wget http://download.vicidial.com/asterisk-patches/app_waitforsilence.c
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mv -f app_waitforsilence.c apps/app_waitforsilence.c
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* REQUIRED for gcc 4.2 systems*
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wget http://download.vicidial.com/asterisk-patches/1.2-gsm-gcc4.2.patch
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patch -p1 ./codecs/gsm/Makefile 1.2-gsm-gcc4.2.patch
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*OPTIONAL* shorter enter and leave sounds for meetme
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mkdir /usr/src/asterisk-1.4
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cd /usr/src/asterisk-1.4
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wget http://downloads.digium.com/pub/asterisk/old-releases/asterisk-1.4.21.2.tar.gz
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wget http://downloads.digium.com/pub/zaptel/zaptel-1.4.12.1.tar.gz
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wget http://downloads.digium.com/pub/libpri/libpri-1.4.9.tar.gz
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tar xzf asterisk-1.4.21.2.tar.gz
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tar xzf zaptel-1.4.12.1.tar.gz
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tar xzf libpri-1.4.9.tar.gz
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cd libpri-1.4.9
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make clean; make; make install
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cd ../zaptel-1.4.12.1
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./configure; make clean; make; make install
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cd ../asterisk-1.4.21.2
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wget http://www.eflo.net/files/enter.h
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wget http://www.eflo.net/files/leave.h
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mv -f enter.h apps/enter.h
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mv -f leave.h apps/leave.h
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wget http://download.vicidial.com/asterisk-patches/1.4-gsm-gcc4.2.patch
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patch -p1 ./codecs/gsm/Makefile 1.4-gsm-gcc4.2.patch
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wget http://download.vicidial.com/conf/res_agi_defunct.patch
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patch -p1 < res_agi_defunct.patch
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File to patch: res/res_agi.c
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cd apps/
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rm -f app_waitforsilence.c
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wget http://download.vicidial.com/conf/app_waitforsilence.c
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cd ../channels/
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rm chan_sip.c
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wget http://download.vicidial.com/conf/chan_sip.c
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cd ../
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./configure; make clean; make; make install
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# this makes sample conf files (only use for new installs)
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make samples
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modprobe zaptel
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modprobe ztdummy (if you have no timer)
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make clean
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make
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make install
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make samples # this makes sample conf files (only use for new installs)
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modprobe zaptel # this loads the zaptel module
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- install the module for the digium device that you are using, we are
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using the T100P single span T1 card so we use:
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modprobe wct1xxp
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@@ -158,6 +122,10 @@ Digium/Clone T100P EXAMPLE, also works for TE4xxp cards
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PHASE 2: CONFIGURING ASTERISK
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### Mostly depricated, ViciDial now allows you to configure phones in the admin
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web interface
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Sample conf files for a system with:
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- astGUIclient and VICIDIAL
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- two SIP phones
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@@ -560,13 +528,11 @@ NEW and submit. Now your system is ready to dial.
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PHASE 8: astGUIclient and VICIDIAL WEB CLIENTS
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The browser requirements for the web-client interfaces are:
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- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
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- Firefox 1.0 or greater *** RECOMMENDED***
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- Mozilla 1.7 or greater
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- Netscape 8 or greater
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- Opera 8.5 or greater
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- Microsoft Internet Explorer 5.0 or greater (we have noticed some serious
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memory leaks in IE5 and IE6 when using this program, they do not occur in
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Firefox which is why we recommend it overall.)
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- Microsoft Internet Explorer 6.0 or greater
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To log into this app you will need a login setup in the vicidial_users table
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as well as an entry for the phone you are using in the phones table. You will
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@@ -336,7 +336,8 @@ make install
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#### OPTIONAL NEWEST VERSIONS OF ASTERISK ####
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#### OPTIONAL 1.2.X VERSIONS OF ASTERISK ####
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(1.4.X instructions are further down)
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mkdir /usr/src/asterisk
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cd /usr/src/asterisk
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@@ -511,34 +512,37 @@ rm -f /usr/lib/asterisk/modules/res_speech.so
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mkdir /usr/src/asterisk-1.4
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cd /usr/src/asterisk-1.4
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wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.21.2.tar.gz
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wget http://downloads.digium.com/pub/zaptel/releases/zaptel-1.4.12.1.tar.gz
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wget http://downloads.digium.com/pub/libpri/releases/libpri-1.4.7.tar.gz
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gunzip asterisk-1.4.21.2.tar.gz
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tar xvf asterisk-1.4.21.2.tar
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gunzip zaptel-1.4.12.1.tar.gz
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tar xvf zaptel-1.4.12.1.tar
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gunzip libpri-1.4.7.tar.gz
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tar xvf libpri-1.4.7.tar
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cd ./zaptel-1.4.12.1
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./configure
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make
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make install
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cd ../libpri-1.4.7
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make clean
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make
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make install
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wget http://downloads.digium.com/pub/asterisk/old-releases/asterisk-1.4.21.2.tar.gz
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wget http://downloads.digium.com/pub/zaptel/zaptel-1.4.12.1.tar.gz
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wget http://downloads.digium.com/pub/libpri/libpri-1.4.9.tar.gz
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tar xzf asterisk-1.4.21.2.tar.gz
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tar xzf zaptel-1.4.12.1.tar.gz
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tar xzf libpri-1.4.9.tar.gz
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cd libpri-1.4.9
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make clean; make; make install
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cd ../zaptel-1.4.12.1
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./configure; make clean; make; make install
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cd ../asterisk-1.4.21.2
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wget http://download.vicidial.com/asterisk-patches/1.4-gsm-gcc4.2.patch
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patch -p1 ./codecs/gsm/Makefile 1.4-gsm-gcc4.2.patch
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wget http://www.eflo.net/files/enter.h
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wget http://www.eflo.net/files/leave.h
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mv -f enter.h apps/enter.h
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mv -f leave.h apps/leave.h
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./configure
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make
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make install
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wget http://download.vicidial.com/asterisk-patches/1.4-gsm-gcc4.2.patch
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patch -p1 ./codecs/gsm/Makefile 1.4-gsm-gcc4.2.patch
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wget http://download.vicidial.com/conf/res_agi_defunct.patch
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patch -p1 < res_agi_defunct.patch
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File to patch: res/res_agi.c
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cd apps/
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rm -f app_waitforsilence.c
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wget http://download.vicidial.com/conf/app_waitforsilence.c
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cd ../channels/
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rm chan_sip.c
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wget http://download.vicidial.com/conf/chan_sip.c
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cd ../
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./configure; make clean; make; make install
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make samples
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modprobe zaptel
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modprobe ztdummy
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+88
-382
@@ -1,4 +1,4 @@
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Asterisk/astguiclient install from scratch. v.2.2.0 2009-04-03
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Asterisk/astguiclient install from scratch. v.2.2.0 2009-06-28
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By the VICIDIAL group info@vicidial.com
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**** IMPORTANT - In order for vicidial/astguiclient to function correctly please
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@@ -6,6 +6,11 @@ read the REQUIREMENTS.txt for a minimum requirements list. ***
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End-user Manuals for Agents and Managers are available from http://www.eflo.net
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||||
|
||||
There is also an easier Ubuntu install doc: Ubuntu_Install.txt
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||||
|
||||
If you want to skip the experience of installing step-by-step completely, take
|
||||
a look at our ISO installer: http://www.vicibox.com
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||||
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||||
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This document is meant to be a very in-depth step-by-step explanation of
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installing the Asterisk open-source PBX on a Linux system and also installing
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@@ -15,46 +20,40 @@ in choices of hardware and software.
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The actual installation that I am doing as I write these instructions will be on
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the following hardware:
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(Changed to more realistic modern hardware)
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- Intel Core 2 Quad CPU
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- SuperMicro Motherboard
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- 4GB RAM
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- 160GB Western Digital SATA ES drive
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- Sangoma VoiceTime USB hardware timer
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- 2U rackmount case with 450W power supply
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- VOIP Phone will be a Zoiper softphone because it's free and does both IAX/SIP
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||||
- Pentium 3 500MHz
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- Intel motherboard D815BN
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- 256MB PC133 RAM
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||||
- 80GB IBM deskstar 7200RPM Hard Drive
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||||
- Digium Wildcard Single Span T1 Card T100P
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||||
- 2U rackmount case with 250W power supply
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- Phone hardware will be a Grandstream BT102 and a Sipura SPA-2000 because they
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are so cheap and readily available
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All of these parts cost less than $1500 together.
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All of these parts, aside from the Digium card and the two SIP VOIP devices,
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||||
were purchased from ebay and the entire package(with the two VOIP devices and
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all server hardware included) cost me about $1100 to put together including the
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phones and Digium adapter.
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||||
We have many other Asterisk servers at our main office, but this one can be
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experimented with easily because it was so cheap to make and has a relatively
|
||||
small capacity when compared with a multi-processor server with a quad span T1
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card.
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||||
|
||||
This is our test Asterisk server and functions well for a dozen or so extensions
|
||||
This is our test Asterisk server and functions well for 100 or so extensions
|
||||
in use if it were to be used in production. A size that is optimal for many
|
||||
small offices operating with a fractional data/voice T1 for instance.
|
||||
small offices operating with a fractional data/voice T1 for instance. This
|
||||
hardware would also be good for a stand-alone ViciDial system with about 50
|
||||
agents handling inbound calls with full recording, or 25 agents doing heavy
|
||||
predictive outbound dialing.
|
||||
|
||||
For hardware you can use almost any Pentium-class processor(PII, PIII, Athlon,
|
||||
Xeon, etc), and you can use any digium telco interface card. Both of these
|
||||
choices will determine what the capacity of your Asterisk server will be. If you
|
||||
want to do simple IVR or conference calling and a few extensions, then a PIII
|
||||
with a single Digium T1 card will work just fine for you. If you want to use the
|
||||
VICIDIAL application, you will want to get as high-powered of a machine as you
|
||||
can afford and get a digium quad-span T1 card.
|
||||
For hardware you can use almost any modern CPU processor(Intel core2, Xeon, etc)
|
||||
, and you can use any Digium or Sangoma telco interface card or USB timer. Both
|
||||
of these choices will determine what the capacity of your Asterisk/ViciDial
|
||||
server will be. If you want to do simple IVR or conference calling and a few
|
||||
extensions, then an Intel Celeron 2GHz with a single Sangoma USB VoiceTime
|
||||
dongle will work just fine for you. If you want to use the ViciDial applications
|
||||
, you will usually want to get as high-powered of a machine as you can afford.
|
||||
|
||||
The following is assumed for these installation procedures:
|
||||
- You have access to a CD burner and 3 blank CDs
|
||||
- You have access to a CD burner and 3 blank CDs(or 1 CD and 1 DVD)
|
||||
- You have some sort of broadband internet connection
|
||||
- You understand basic linux commands and can use a file editor like vi
|
||||
- You have all of the necessary hardware:
|
||||
- a pentium-class computer
|
||||
- a digium telco interface card with appropriate telco lines
|
||||
- at least 1 SIP VOIP device
|
||||
- at least 1 SIP/IAX VOIP device(this can be a softphone too)
|
||||
- a Local Area Network(LAN) with extra ports enough for the new server
|
||||
and the number of phones you want
|
||||
|
||||
@@ -65,7 +64,7 @@ and the number of phones you want
|
||||
PHASE 1: INSTALLING AN OPERATING SYSTEM
|
||||
|
||||
|
||||
This installation will be using Slackware 11 for the linux distribution,
|
||||
This installation will be using Slackware 12 for the linux distribution,
|
||||
Slackware 12.X will also work with these instructions. There are several easier
|
||||
linux distributions and there are others that are more popular, but Slackware is
|
||||
a nice non-commercial distro that has been around for a long time and proven
|
||||
@@ -83,7 +82,7 @@ to it. You will select "filesystem utilities" and then "XFDISK"
|
||||
|
||||
3. Select any old partitions and delete them and then create 2 new
|
||||
partitions:
|
||||
- 70000 MB, select yes to validate, change partition type to "Linux Native"
|
||||
- 150000 MB, select yes to validate, change partition type to "Linux Native"
|
||||
- 3332 MB, select yes to validate, change partition type to "Linux Swap"
|
||||
- press F3 to exit and let it do it's thing, this will take an hour or so.
|
||||
|
||||
@@ -172,10 +171,9 @@ If you are using a newer Digium Octasic-based echo-cancellation quad T1/E1 card
|
||||
then you need to use a 2.6 Linux kernel in order to use the echo-cancellation
|
||||
functionality of the card.
|
||||
|
||||
If you want to compile a 2.6 kernel then start with OPTION 1, otherwise to
|
||||
compile the 2.4 kernel that comes with Slackware(2.4.33) start with OPTION 2:
|
||||
If you want to compile a 2.6 kernel then start with OPTION 1:
|
||||
|
||||
OPTION 1: compile Linux kernel 2.6.17 *RECOMMENDED*
|
||||
OPTION 1: compile Linux kernel 2.6.17 *NOT REQUIRED*
|
||||
|
||||
1. cd /usr/src
|
||||
2. wget http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.17.11.tar.gz
|
||||
@@ -273,44 +271,6 @@ OPTION 1: compile Linux kernel 2.6.17 *RECOMMENDED*
|
||||
tmpfs /var/spool/asterisk/monitor tmpfs rw 0 0
|
||||
21. shutdown -r 0 # reboot machine and hope it worked
|
||||
|
||||
OPTION 2: compile Linux kernel 2.4.33.3 (not recommended, very old)
|
||||
|
||||
1. cd /usr/src/linux # move to your linux source directory
|
||||
2. cp .config config.save # copy old config to a save file
|
||||
3. make mrproper # prep for kernel assembly
|
||||
4. make menuconfig # launch configuration menu app
|
||||
(this part is very dependant upon your own hardware)
|
||||
enable processor version # select the processor that you have
|
||||
enable SMP # if more than 1 processor or Intel HT
|
||||
enable high memory () # if more than 1GB of RAM
|
||||
enable SCSI Multiple # if SCSI drives
|
||||
enable SCSI devices AMI Megaraid # if SCSI Megaraid adapter
|
||||
enable 3com network devices # if 3com network card
|
||||
enable ext3 file system # for ext3 to work
|
||||
enable all ACPI options # for SMP to work
|
||||
enable Enhanced Real Time Clock Support in Character devices section
|
||||
# for SMP to work
|
||||
enable any other hardware specific options
|
||||
exit and save configuration
|
||||
5. make dep # build the kernel dependancies
|
||||
6. make clean # clean up the kernel build areas
|
||||
7. make bzImage # create a kernel bzImage
|
||||
8. make modules # build the modules into the image
|
||||
9. make modules_install # install kernel modules
|
||||
10. # nothing# mkinitrd /boot/initrd-XXXXXX.img XXXXXX *not needed on Slackware*
|
||||
11. cp arch/i386/boot/bzImage /boot/bzImage-XXXXXX # copy image
|
||||
(put whatever you want in XXXXXX, that is your new kernel name)
|
||||
12. cp System.map /boot/System.map-XXXXXXN # copy system map
|
||||
13. mv -f /boot/System.map /boot/System.map-orig
|
||||
14. ln -s /boot/System.map-XXXXXX /boot/System.map # symlink map
|
||||
15. vi /etc/lilo.conf # edit the lilo boot config file
|
||||
image=/boot/bzImage-XXXXXX # add the new image in above-
|
||||
label=test-XXXXXX # the previous one
|
||||
root=/dev/hda1 # device of root partition
|
||||
read-only
|
||||
16. /sbin/lilo # run the lilo reload script
|
||||
17. shutdown -r 0 # reboot machine and hope it worked
|
||||
|
||||
|
||||
After compiling your kernel you can run a few commands to verify that you are
|
||||
running your new kernel and that devices are running as they are supposed to:
|
||||
@@ -397,6 +357,7 @@ a more loaded system.
|
||||
skip-name-resolve
|
||||
# comment out the line 'log-bin=mysql-bin'
|
||||
max_connections = 200
|
||||
connect_timeout=60
|
||||
|
||||
**** For some systems you may need to add the mysql/bin directory to your PATH:
|
||||
- PATH=$PATH:$HOME/bin:/usr/local/mysql/bin/
|
||||
@@ -423,12 +384,10 @@ a more loaded system.
|
||||
|
||||
SUBPHASE 3.2: Installing Perl Modules
|
||||
|
||||
NOTE - you can install ActiveState http://www.activestate.com perl which may
|
||||
improve performance, but it is not required. Here's the source for ActiveState
|
||||
NOTE - you can install ActiveState http://www.activestate.com perl, but it is
|
||||
not required. Here's the source for ActiveState
|
||||
Perl 5.8: (it's free)
|
||||
http://downloads.activestate.com/ActivePerl/src/5.8/AP817_source.tgz
|
||||
I hope to add the lengthy steps for installing it as your default perl on your
|
||||
server but I need some time and a free machine to do that.
|
||||
|
||||
cpan is the "Comprehensive Perl Archive Network". It's a mirrored archive of
|
||||
most of the perl modules out there complete with a installation and management
|
||||
@@ -475,7 +434,7 @@ following modules first: (say YES if asked to install prerequisites)
|
||||
- then quit cpan, you are done
|
||||
5. Go to http://asterisk.gnuinter.net/ and download the asterisk-perl module
|
||||
(backup link: http://download.vicidial.com/packages/asterisk-perl-0.08.tar.gz)
|
||||
NOTE: Do NOT use the 0.09 version, it does not work with VICIDIAL
|
||||
NOTE: Do NOT use the 0.09 or any newer version, they do not work with VICIDIAL
|
||||
- to install this directly on the command line type:
|
||||
- cd /usr/local
|
||||
- wget http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz
|
||||
@@ -886,121 +845,54 @@ PHASE 4: INSTALLING ASTERISK
|
||||
|
||||
OK, all the prep work is done, now it's time to start having fun with Asterisk.
|
||||
There are two basic ways to install Asterisk, an official release(at the time of
|
||||
this writing the official release is 1.2.30.2) and the SVN_DEV version(development
|
||||
branch). We recommend using Asterisk 1.2.30.2. These instructions are how we get
|
||||
our Asterisk system with it's T1 line installed with our 2 SIP VOIP devices and
|
||||
one IAX2 softphone.
|
||||
this writing the recommended release is 1.4.21.2). These instructions are how we
|
||||
get our Asterisk system with it's T1 line installed with our Zoiper softphone.
|
||||
|
||||
NOTE: If you want to use Asterisk 1.4, you will need to make sure that you set
|
||||
NOTE: If you want to use Asterisk 1.2, you will need to make sure that you set
|
||||
the servers table "asterisk_version" field to the proper version number and you
|
||||
can use the docs/conf_examples/extensions.conf.sample-1.4 file for your default
|
||||
can use the docs/conf_examples/extensions.conf.sample file for your default
|
||||
dialplan
|
||||
|
||||
NOTE: If you want to use release 1.0.8 or 9 we would recommend either using the
|
||||
CVS_v1-0 branch where the issues are fixed, or patching your 1.0.8/1.0.9 code
|
||||
with the following patch:
|
||||
(http://astguiclient.sourceforge.net/experimental_code/localmasq.patch)
|
||||
- If you do patch your system make sure you put the asterisk version
|
||||
field for the server on the admin pages as '1.0.11.1'
|
||||
|
||||
1. follow these command line steps:
|
||||
- mkdir /usr/src/asterisk
|
||||
- cd /usr/src/asterisk
|
||||
A. if you want 1.2 release (reliable with new features):
|
||||
- wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.2.30.2.tar.gz
|
||||
- wget http://downloads.digium.com/pub/zaptel/releases/zaptel-1.2.27.tar.gz
|
||||
- wget http://downloads.digium.com/pub/libpri/releases/libpri-1.2.5.tar.gz
|
||||
- gunzip asterisk-1.2.30.2.tar.gz
|
||||
- tar xvf asterisk-1.2.30.2.tar
|
||||
- gunzip zaptel-1.2.27.tar.gz
|
||||
- tar xvf zaptel-1.2.27.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
|
||||
- svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2
|
||||
C. if you want latest SVN_DEV version (not recommended) [1.6 tree]
|
||||
- svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
|
||||
- svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
|
||||
- svn checkout http://svn.digium.com/svn/libpri/trunk libpri
|
||||
- ALL ->
|
||||
- (1.0 tree)if you want to allow for more than 100 voicemail messages in a
|
||||
mailbox(warning this will slightly increase memory usage when a call
|
||||
is in voicemail) edit the voicemail source code file:
|
||||
- vi /usr/src/asterisk/asterisk/apps/app_voicemail.c
|
||||
edit this line and change 100 to 999:
|
||||
#define MAXMSG 100
|
||||
- (1.0 tree)if you have no X server installed on your Asterisk machine, then you
|
||||
will need to comment out the gtk console lib(only affects 1.0 releases)
|
||||
edit the voicemail source code file:
|
||||
- vi /usr/src/asterisk/asterisk/pbx/Makefile
|
||||
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-1.2.27
|
||||
- make clean
|
||||
- make
|
||||
- make install
|
||||
- cd ../libpri-1.2.5
|
||||
- make clean
|
||||
- make
|
||||
- make install
|
||||
- cd ../asterisk-1.2.30.2
|
||||
- (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
|
||||
- cd apps
|
||||
- wget http://www.eflo.net/files/app_amd2.c
|
||||
- mv app_amd2.c app_amd.c
|
||||
- vi Makefile
|
||||
replace this line(line 32):
|
||||
app_mixmonitor.so app_stack.so
|
||||
with this line:
|
||||
app_mixmonitor.so app_stack.so app_amd.so
|
||||
- wget http://www.eflo.net/files/amd2.conf
|
||||
- mkdir /etc/asterisk
|
||||
- mv amd2.conf /etc/asterisk/amd.conf
|
||||
*OPTIONAL*(1.2.23 thru 1.2.30.2) apply the meetme DTMF passthru patch
|
||||
- wget http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch
|
||||
- patch -p1 < ./meetme_DTMF_passthru-1.2.23.patch
|
||||
- File to patch: app_meetme.c
|
||||
*OPTIONAL*(1.2.12.1 thru 1.2.30.2) 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
|
||||
- File to patch: app_meetme.c
|
||||
- cd ../
|
||||
|
||||
-(1.2 tree) apply the cli delimiter patch
|
||||
- wget http://www.eflo.net/files/cli_chan_concise_delimiter.patch
|
||||
- patch -p1 < ./cli_chan_concise_delimiter.patch
|
||||
- File to patch: cli.c
|
||||
|
||||
-(gcc version 4.2) apply the gsm audio codec patch to fix gsm
|
||||
- wget http://download.vicidial.com/asterisk-patches/1.2-gsm-gcc4.2.patch
|
||||
- patch -p1 ./codecs/gsm/Makefile 1.2-gsm-gcc4.2.patch
|
||||
|
||||
*OPTIONAL*(1.2.14 thru 1.2.30.2) rewrite of waitforsilence
|
||||
- wget http://download.vicidial.com/asterisk-patches/app_waitforsilence.c
|
||||
- mv -f app_waitforsilence.c apps/app_waitforsilence.c
|
||||
|
||||
*OPTIONAL* shorter enter and leave sounds for meetme
|
||||
- wget http://www.eflo.net/files/enter.h
|
||||
- wget http://www.eflo.net/files/leave.h
|
||||
- mv -f enter.h apps/enter.h
|
||||
- mv -f leave.h apps/leave.h
|
||||
|
||||
|
||||
- make clean
|
||||
- make
|
||||
- make install
|
||||
- make samples # this makes sample conf files (only use for new installs)
|
||||
mkdir /usr/src/asterisk-1.4
|
||||
cd /usr/src/asterisk-1.4
|
||||
wget http://downloads.digium.com/pub/asterisk/old-releases/asterisk-1.4.21.2.tar.gz
|
||||
wget http://downloads.digium.com/pub/zaptel/zaptel-1.4.12.1.tar.gz
|
||||
wget http://downloads.digium.com/pub/libpri/libpri-1.4.9.tar.gz
|
||||
tar xzf asterisk-1.4.21.2.tar.gz
|
||||
tar xzf zaptel-1.4.12.1.tar.gz
|
||||
tar xzf libpri-1.4.9.tar.gz
|
||||
cd libpri-1.4.9
|
||||
make clean; make; make install
|
||||
cd ../zaptel-1.4.12.1
|
||||
./configure; make clean; make; make install
|
||||
cd ../asterisk-1.4.21.2
|
||||
wget http://www.eflo.net/files/enter.h
|
||||
wget http://www.eflo.net/files/leave.h
|
||||
mv -f enter.h apps/enter.h
|
||||
mv -f leave.h apps/leave.h
|
||||
wget http://download.vicidial.com/asterisk-patches/1.4-gsm-gcc4.2.patch
|
||||
patch -p1 ./codecs/gsm/Makefile 1.4-gsm-gcc4.2.patch
|
||||
wget http://download.vicidial.com/conf/res_agi_defunct.patch
|
||||
patch -p1 < res_agi_defunct.patch
|
||||
File to patch: res/res_agi.c
|
||||
cd apps/
|
||||
rm -f app_waitforsilence.c
|
||||
wget http://download.vicidial.com/conf/app_waitforsilence.c
|
||||
cd ../channels/
|
||||
rm chan_sip.c
|
||||
wget http://download.vicidial.com/conf/chan_sip.c
|
||||
cd ../
|
||||
./configure; make clean; make; make install
|
||||
# this makes sample conf files (only use for new installs)
|
||||
make samples
|
||||
modprobe zaptel
|
||||
|
||||
- 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:
|
||||
- modprobe wct1xxp
|
||||
or if you have no timer, just "modprobe ztdummy"
|
||||
Here's the list of all digium cards and the modules you use with
|
||||
them:
|
||||
Card Module
|
||||
@@ -1429,141 +1321,8 @@ files, so be warned:
|
||||
- vi /etc/asterisk/extensions.conf
|
||||
|
||||
You should be using the sample extensions.conf that is included with the
|
||||
release of VICIDIAL that you installed, below is just an explanation of what
|
||||
most of those entries do and why they are there.
|
||||
|
||||
This is known as the dialplan. Since we are installing a
|
||||
Long-Distance T1 with one 800 number on it, we will need to put that
|
||||
800 number in the plan, as well as how to dial out through the T1
|
||||
lines and we will need to add an entry for each of the phones that we
|
||||
have just set up in the sip.conf file. There are many examples both
|
||||
in the sample file and online for what to put in your dialplan, here
|
||||
is the simplified dialplan that we are using:
|
||||
######------ START extensions.conf example ------######
|
||||
[general]
|
||||
static=yes
|
||||
writeprotect=no
|
||||
|
||||
[globals]
|
||||
CONSOLE=Console/dsp ; Console interface for demo
|
||||
TRUNK=Zap/g1 ; Trunk interface
|
||||
TRUNKX=Zap/g2 ; 2nd trunk interface
|
||||
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
|
||||
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
|
||||
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing
|
||||
|
||||
[default]
|
||||
; Extension 8600 + 8601 conference rooms
|
||||
exten => 8600,1,Meetme,8600
|
||||
exten => 8601,1,Meetme,8601
|
||||
|
||||
; Extension 102 - Grandstream hardphone
|
||||
exten => 102,1,Playback,transfer|skip ; "Please hold while..."
|
||||
exten => 102,2,Dial,sip/gs102|20|to ; Ring, 20 secs max
|
||||
exten => 102,3,Voicemail,u102 ; Send to voicemail...
|
||||
; Extension 2000 Sipura line 1
|
||||
exten => 2000,1,Dial,sip/spa2000|30|to ; Ring, 30 secs max
|
||||
exten => 2000,2,Voicemail,u2000 ; Send to voicemail...
|
||||
; Extension 2001 Sipura line 2
|
||||
exten => 2001,1,Dial,sip/spa2001|30|to ; Ring, 30 secs max
|
||||
exten => 2001,2,Voicemail,u2001 ; Send to voicemail...
|
||||
; Extension 2020 rings both sipura lines
|
||||
exten => 2001,1,Dial,sip/spa2000&sip/spa2001|30|to ; Ring, 30 secs max
|
||||
exten => 2001,2,Voicemail,u2000 ; Send to voicemail...
|
||||
; Extension 301 rings the firefly softphone
|
||||
exten => 301,1,Dial,(IAX2/firefly01@firefly01/s)
|
||||
exten => 301,2,Hangup
|
||||
|
||||
; Extension 3429 - Inbound 800 number (1-800-555-3429)
|
||||
exten => _**3429,1,Ringing
|
||||
exten => _**3429,2,Answer
|
||||
exten => _**3429,3,Dial,sip/spa2000&sip/spa2001|30|to ; Ring, 30 secs max
|
||||
exten => _**3429,4,Voicemail,u2000 ; Send to voicemail...
|
||||
; Extension 3429 - with ANI [callerID]
|
||||
exten => _*NXXNXXXXXX*3429,1,Ringing
|
||||
exten => _*NXXNXXXXXX*3429,2,Answer
|
||||
exten => _*NXXNXXXXXX*3429,3,Dial,sip/spa2000&sip/spa2001|30|to ; Ring, 30
|
||||
secs max
|
||||
exten => _*NXXNXXXXXX*3429,4,Voicemail,u2000 ; Send to voicemail...
|
||||
|
||||
; dial a long distance outbound number to the UK
|
||||
exten => _901144XXXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},55,tTo)
|
||||
exten => _901144XXXXXXXXXX,2,Hangup
|
||||
|
||||
; dial a long distance outbound number to Australia
|
||||
exten => _901161XXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,tTo)
|
||||
exten => _901161XXXXXXXXX,2,Hangup
|
||||
|
||||
; dial an 800 outbound number
|
||||
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
|
||||
exten => _91800NXXXXXX,2,Hangup
|
||||
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
|
||||
exten => _91888NXXXXXX,2,Hangup
|
||||
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
|
||||
exten => _91877NXXXXXX,2,Hangup
|
||||
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
|
||||
exten => _91866NXXXXXX,2,Hangup
|
||||
|
||||
; dial a local 727 outbound number with area code
|
||||
exten => _9727NXXXXXX,1,Dial(${TRUNK}/1${EXTEN:1},,tTo)
|
||||
exten => _9727NXXXXXX,2,Hangup
|
||||
|
||||
; dial a local 813 outbound number with area code
|
||||
exten => _9813NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
|
||||
exten => _9813NXXXXXX,2,Hangup
|
||||
|
||||
; dial a long distance outbound number
|
||||
exten => _91NXXNXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,tTo)
|
||||
exten => _91NXXNXXXXXX,2,Hangup
|
||||
|
||||
; dial a local outbound number (modified because of only LD T1)
|
||||
exten => _9NXXXXXX,1,Dial(${TRUNK}/1727${EXTEN:1},,tTo)
|
||||
exten => _9NXXXXXX,2,Hangup
|
||||
|
||||
; barge monitoring extension
|
||||
exten => 8159,1,ZapBarge
|
||||
exten => 8159,2,Hangup
|
||||
|
||||
; # timeout invalid rules
|
||||
exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
|
||||
exten => #,2,Hangup ; Hang them up.
|
||||
exten => t,1,Goto(#,1) ; If they take too long, give up
|
||||
exten => i,1,Playback(invalid) ; "That's not valid, try again"
|
||||
|
||||
; Give voicemail at extension 8500
|
||||
exten => 8500,1,VoicemailMain
|
||||
exten => 8500,2,Goto(s,6)
|
||||
|
||||
; ASTERISK AGENTS LOGINS FOR QUEUES (NOT part of VICIDIAL)
|
||||
; the following assumes phone agent login and exten are 3 digits and the same
|
||||
; also assumes that 3-digit login is present in agents.conf and queueus.conf
|
||||
;Agent Logout then stay onhook, DIAL 54 + 3-digit ID
|
||||
exten => _54XXX,1,AgentCallbackLogin(||)
|
||||
; the following are used to login and logout of Asterisk Queues from phone
|
||||
;Agent Login then stay offhook on the phone, DIAL 55 + 3-digit ID
|
||||
exten => _55XXX,1,AgentLogin(${EXTEN:1})
|
||||
;Agent Login then stay onhook, phones will ring, DIAL 56 + 3-digit ID
|
||||
exten => _56XXX,1,AgentCallbackLogin(||${EXTEN:1}@default)
|
||||
|
||||
######------ END extensions.conf example ------######
|
||||
|
||||
### OPTIONAL IAX trunk extensions entries for long distance dialing over IAX
|
||||
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
|
||||
exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,o)
|
||||
exten => _91NXXNXXXXXX,3,Hangup
|
||||
|
||||
### OPTIONAL SIP trunk extensions entries for long distance dialing over SIP
|
||||
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
|
||||
exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)
|
||||
exten => _91NXXNXXXXXX,3,Hangup
|
||||
|
||||
### OPTIONAL IAX Load Balance extens to allow for Overflow and Balanced VDAD
|
||||
### In this setup, the serverIP is the prefix followed by agent conf_exten
|
||||
### FOR MORE INFORMATION, READ THE LOAD_BALANCING.txt DOCUMENT
|
||||
### server 1 extens:
|
||||
exten => _010*010*010*016*.,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)
|
||||
### server 2 extens:
|
||||
exten => _010*010*010*015*.,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
|
||||
release of VICIDIAL that you installed. There is also an auto-generated
|
||||
extensions-vicidial.conf file that contains dynamic dialplan entries.
|
||||
|
||||
10. edit dnsmgr.conf: # It is very helpful to enable dnsmgr
|
||||
[general]
|
||||
@@ -1781,7 +1540,7 @@ create database asterisk;
|
||||
|
||||
NOTE: if you will be using lead files with a language that does not use the
|
||||
standard latin character set then you will want to use UTF8 for your default
|
||||
characterset in the MySQL database. This requires at least MySQL 4.1.11 and you
|
||||
characterset in the MySQL database. This requires at least MySQL 5.0.X and you
|
||||
can use the following query to create the database:
|
||||
CREATE DATABASE `asterisk` DEFAULT CHARACTER SET utf8 COLLATE utf8_unicode_ci;
|
||||
|
||||
@@ -1794,9 +1553,6 @@ GRANT RELOAD ON *.* TO cron@localhost;
|
||||
|
||||
flush privileges;
|
||||
|
||||
# NOTE: if using MySQL 4.1.X or higher(not 5.X) you may need to run this query too:
|
||||
UPDATE mysql.user set password=OLD_PASSWORD('1234') where user='cron';
|
||||
|
||||
# To make sure that new processes can connect to the Database under load we should
|
||||
# increase the global connect_timeout
|
||||
|
||||
@@ -1909,7 +1665,9 @@ to work with it.
|
||||
|
||||
Again, if you have selected to use the sample conf files during installation
|
||||
then you do not have to add any of these lines to your conf files, they should
|
||||
already be included.
|
||||
already be included. There is also an auto-generated
|
||||
extensions-vicidial.conf file that contains dynamic dialplan entries.
|
||||
|
||||
|
||||
1. Add the call_log entries to all incoming/outgoing extensions entries:
|
||||
- here is how our sample dialplan changes for adding call_log entries(only
|
||||
@@ -2009,7 +1767,9 @@ exten => _9NXXXXXX,3,Hangup
|
||||
|
||||
|
||||
2. Add the call_inbound entries to all incoming extensions entries that you want
|
||||
CallerID popups on:
|
||||
CallerID popups on(ONLY WORKS IN astguiclient.php APPLICATION):
|
||||
(Using the DID configuration is the preferred method of call routing in ViciDial)
|
||||
|
||||
- here is how our sample dialplan changes for adding call_inbound
|
||||
entries(only effected extension groups are show):
|
||||
######------ START extensions.conf changes for call_inbound ------######
|
||||
@@ -2034,61 +1794,6 @@ exten => _*NXXNXXXXXX*3429,3,Answer
|
||||
exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to
|
||||
exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000
|
||||
|
||||
; parameters for agi-VDAD_ALL_inbound.agi (9 fields separated by five dashes "-----"):
|
||||
; 1. the method of call handling for the script:
|
||||
; - CID - CID received, add record with phone number
|
||||
; - CIDLOOKUP - Lookup CID to find record in whole system
|
||||
; - CIDLOOKUPRL - Restrict lookup to one list
|
||||
; - CIDLOOKUPRC - Restrict lookup to one campaign's lists
|
||||
; - CLOSER - Closer calls from VICIDIAL fronters
|
||||
; - ANI - ANI received, add record with phone number
|
||||
; - ANILOOKUP - Lookup ANI to find record in whole system
|
||||
; - ANILOOKUPRL - Restrict lookup to one list
|
||||
; - 3DIGITID - Enter 3 digit code to go to agent
|
||||
; - 4DIGITID - Enter 4 digit code to go to agent
|
||||
; - 5DIGITID - Enter 5 digit code to go to agent
|
||||
; - 10DIGITID - Enter 10 digit code to go to agent
|
||||
; 2. the method of searching for an available agent:
|
||||
; - LO - Load Balance Overflow only (priority to home server)
|
||||
; - LB - <default> Load Balance total system
|
||||
; - SO - Home server only
|
||||
; 3. the full name of the IN GROUP to be used in vicidial for the inbound call
|
||||
; 4. the phone number that was called, for the log entry
|
||||
; 5. the callerID or lead_id of the person that called(usually overridden)
|
||||
; 6. the park extension audio file name if used
|
||||
; 7. the status of the call initially(usually not used)
|
||||
; 8. the list_id to insert the new lead under if it is new (and CID/ANI available)
|
||||
; 9. the phone dialing code to insert with the new lead if new (and CID/ANI available)
|
||||
; 10. the campaign_id to search within lists if CIDLOOKUPRC
|
||||
; inbound VICIDIAL call with CID delivery through T1 PRI
|
||||
exten => 1234,1,Answer ; Answer the line
|
||||
exten => 1234,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1)
|
||||
exten => 1234,3,Hangup
|
||||
|
||||
; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
|
||||
exten => _90009.,1,Answer ; Answer the line
|
||||
exten => _90009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
|
||||
exten => _90009.,3,Hangup
|
||||
exten => _990009.,1,Answer ; Answer the line
|
||||
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
|
||||
exten => _990009.,3,Hangup
|
||||
|
||||
|
||||
### follow these instructions if you plan to have VICIDIAL agents take inbound or closer calls:
|
||||
1. in VICIDIAL web admin "add a new in-group" (the above examples would be "CL_GALLERIA")
|
||||
- group IDs cannot contain spaces ' ' or dashes '-' or plusses '+'
|
||||
- if you are using a HEX color value make sure to include the hash '#' at the beginning
|
||||
2. create a new campaign in VICIDIAL called "CLOSER" and set "allow inbound blended" to Y
|
||||
3. check the CL_GALLERIA checknox in the "Allowed In-Groups" section
|
||||
4. have agents log in to the CLOSER campaign and select the CL_GALLERIA in-group
|
||||
5. they will now start receiving inbound calls
|
||||
6. as calls come in, each call is inserted into the vicidial_list table under the
|
||||
list specified int the AGI string, In the above example that would be list 999
|
||||
7. if you want to take closer calls from the campaign "TEST" you will need to create
|
||||
an in-group called "CL_TEST_" for internal closing(on the same system) or "CL_TEST_L"
|
||||
for local closing(closer on different system from fronter) and then the fronter will
|
||||
click on the "internal closer" button to send the call to a closer
|
||||
|
||||
* NOTE, you need to set the dial_level of the CLOSER campaign to 1 or higher for inbound/closers to work
|
||||
|
||||
######------ END extensions.conf changes for call_inbound ------######
|
||||
@@ -2320,7 +2025,7 @@ exten => _86000[0-4]X,1,Meetme,${EXTEN}|q
|
||||
|
||||
6. Add the conference entries for VICIDIAL conferences:
|
||||
- here is how our sample dialplan changes for adding VICIDIAL conference
|
||||
entries(this is a pure addition, nothing is being modified):
|
||||
entries(this is already in the sample extensions.conf file):
|
||||
NOTE: see below these entries for app_conference instructions is used
|
||||
######------ START extensions.conf changes for VD conf ------######
|
||||
exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
|
||||
@@ -2364,6 +2069,7 @@ NOTE: If you want to do DTMF passthru with app_conference bee sure to add the
|
||||
7. Add the more entries for astGUIclient specific uses:
|
||||
- here are some more dialplan additions needed to use astGUIclient(this is a
|
||||
pure addition, nothing is being modified):
|
||||
(this is already in the sample extensions.conf file)
|
||||
######------ START extensions.conf other additions ------######
|
||||
; park channel for client GUI parking, hangup after 30 minutes
|
||||
; create a GSM formatted audio file named "park.gsm" that is 30 minutes long
|
||||
@@ -2808,7 +2514,7 @@ SUBPHASE 6.8: astGUIclient web-only client:
|
||||
With 1.1.1 release of astguiclient we have completely rewritten the astGUIclient
|
||||
client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time
|
||||
GUI interface using only a web browser. The browser requirements for this are:
|
||||
- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
|
||||
- Firefox 1.0 or greater *** RECOMMENDED ***
|
||||
- Mozilla 1.7 or greater
|
||||
- Netscape 8 or greater
|
||||
- Opera 8.5 or greater
|
||||
@@ -2856,7 +2562,7 @@ With 1.1.6 release of astguiclient we have finished the rewrite of the VICIDIAL
|
||||
client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time
|
||||
GUI interface using only a web browser like we have done with astGUIclient.
|
||||
The browser requirements for this are:
|
||||
- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
|
||||
- Firefox 1.0 or greater *** RECOMMENDED ***
|
||||
- Mozilla 1.7 or greater
|
||||
- Netscape 8 or greater
|
||||
- Opera 8.5 or greater
|
||||
|
||||
@@ -81,7 +81,7 @@ server_id VARCHAR(10) NOT NULL,
|
||||
server_description VARCHAR(255),
|
||||
server_ip VARCHAR(15) NOT NULL,
|
||||
active ENUM('Y','N'),
|
||||
asterisk_version VARCHAR(20) default '1.2.9',
|
||||
asterisk_version VARCHAR(20) default '1.4.21.1',
|
||||
max_vicidial_trunks SMALLINT(4) default '23',
|
||||
telnet_host VARCHAR(20) NOT NULL default 'localhost',
|
||||
telnet_port INT(5) NOT NULL default '5038',
|
||||
|
||||
@@ -1,4 +1,4 @@
|
||||
INSERT INTO servers (server_id,server_description,server_ip,active,asterisk_version)values('TESTast','Test install of Asterisk server', '10.10.10.15','Y','1.2.26.2');
|
||||
INSERT INTO servers (server_id,server_description,server_ip,active,asterisk_version)values('TESTast','Test install of Asterisk server', '10.10.10.15','Y','1.4.21.2');
|
||||
|
||||
INSERT INTO server_updater SET server_ip='10.10.10.15', last_update='';
|
||||
|
||||
|
||||
Reference in New Issue
Block a user