...
First it is needed to install Asterisk. In most distributions, this should be easy because Asterisk is packaged with your distro. So use your distribution package manager to install. This howto is not made to focus on installation step. See 3.1.2. Setting up SIP voice services for an institution with Asterisk
Install HW drivers
You have to be sure that drivers for your ISDN PRI card are working. Simplest way is probably to install zaptel drivers (or it is together with your Asterisk package from your distro). See 3.1.2. Setting up SIP voice services for an institution with Asterisk
Ensure that hardware is working
...
Configure Layer1 and Layer2 parameters
This parameters are setuped set by /etc/zaptel.conf (change xx to your country code (eg. uk or de). Disable crc4 if your operator or PBX does not support it.
| Code Block |
|---|
loadzone = xx
defaultzone = xx
# PRI TE
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-30
|
Next, it is needed to set layer3 parameters and PBX specific parameters. It is stored in /etc/asterisk/zapata.conf. Use same language code as above. Use swtichtype ISDN because it has best support within asterisk. Qsig is not implemented well. Select signalling (either cpe for end device and net for master device). In most situation, your asterisk gateway will act as CPE. Dialplan is unknown, it means that there will be no automatic country or PBX prefixes for calling and we will do this in dialplan (better solution). Echoparaeters are needed for echocancelation to work. But it can be time consuming process to find source of echo and cancelate it. And you can never be sure that it will not appear in some calls. See http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation&view_comment_id=13700 for more informations. Immediate means if calling to your asterisk will be routed to called number into dialplan. Next, we have to setup calling groups. In this scenario, we have only one calling group which spans all available channels on card. We can group only some of them or more cards together. Any call to your Asterisk box will be routed into fpstn context (see above).
| Code Block |
|---|
[channels]
language=xx
switchtype = euroisdn
signalling = pri_cpe
pridialplan = unknown
prilocaldialplan = unknown
echocancel=64
echotraining = 150
echocancelwhenbridged=yes
;txgain=-4
;rxgain=-4
immediate=no
group = 1
context = fpstn
channel => 1-15,17-30
acountcode = fpstn
|