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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 1.2.17) and the SVN_DEV version(development branch). We recommend using Asterisk 1.2.17. 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.
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://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
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) [1.5 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-1.2.16
- make clean
- make
- make install
- cd ../libpri-1.2.4
- make clean
- make
- make install
- cd ../asterisk-1.2.17
- (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*(1.2.12.1 thru 1.2.17) 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
- make clean
- make
- make install
- make samples # this makes sample conf files (only use for new installs)
- (1.2 tree) EXPERIMENTAL!!! 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:
- modprobe wct1xxp
Here's the list of all digium cards and the modules you use with them:
Card Module
-----------------
TDM400P wctdm
X100P wcfxo
TDM* wcfxs
S100U wcusb
T100P wct1xxp
E100P wct1xxp
T400P tor2
E400P tor2
TE110P wcte11xp
TE410P wct4xxp
TE405P wct4xxp
TE411P wct4xxp
TE406P wct4xxp
TE210P wct2xxp
TE205P wct2xxp
TDM2400P wctdm24xxp
- 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-2.3.4-7.tgz
- now your asterisk installation is built and loaded and it's time to configure it.
NOTES: If you want to install zttool diagnostics you may need the newt package installed:
- wget http://www.freedomphones.net/files/newt-0.51.6.tar.gz
- gunzip newt-0.51.6.tar.gz
- tar xvf newt-0.51.6.tar
- cd newt-0.51.6
- ./configure
- make
- make install
- cd ../
- ln -s /usr/lib/libnewt.so.0.51.6 /usr/lib/libnewt.so.0.51
then go to your zaptel folder and do 'make zttool'
Digium/Clone X100P EXAMPLE:
Here is an example of a configuration where an X100P single FXO card is used for zaptel timing and not used for calling:
NOTE: you can get an X100P through ebay for $10-$30 that will work with Asterisk
/etc/zaptel.conf:
loadzone=us
defaultzone=us
fxsks=1
/etc/asterisk/zapata.conf:
[trunkgroups]
[channels]
context=unused
signalling=fxs_ks
channel => 1
Added this to the rc.local file:
# Load zaptel drivers for x100p
modprobe zaptel
modprobe wcfxo
