Setting up Zaptel/Asterisk on Alpine
This material is obsolete ... Please update this guide to a supported version of Alpine. It was written with an Ancient release in mind (Discuss) |
Asterisk is an open-source voip server. It can be used both with sip-clients as with phones and/or phonesystems.
Note:
We used Alpine version 1.1.3-beta8
Installation
Booted from CD
Log in as root, no password needed
$ setup-alpine
Set hostname, network settngs and root-password
$ apk_add openssh asterisk
Make ssh run at bootup:
$ rc_add sshd
Configure asterisk, we copied settings from our previous install, on Debian Sarge. The only change we had to make to our previous asterisk config was:
$ vi modules.conf
under [modules], make sure
load => “res_musiconhold.so”
is loaded before other modules
Make asterisk run at bootup:
$ rc_add asterisk
Zaptel
We use an isdn card to connect to our phone-center. It uses the zaptel driver. To load all the needed modules we had to make the following changes:
$ vi /etc/modules
Add the following modules:
zaphfc zaptel af_packet
$vi /etc/modules.conf
Add:
options torisa base=0xd0000 alias char-major-196 torisa alias wctdm wcfxs alias wct2xxp wct4xxp
Wrapping up
Commit to floppy
$ lbu co floppy
Reboot to make sure you made no mistakes, done.
I prefer to reboot staright after installation. If I made a mistake, I rather find out now then in a couple of monts, when I will for sure have forgotton how I set it all up.
Permissions
To run asterisk as user asterisk, we had to add asterisk to the dialout group:
$ grep asterisk /etc/group
dialout:x:20:root,asterisk
To be able to support sip dial-in clients, we run asterisk as root:
$ vi /etc/conf.d/asterisk
And set:
ASTERISK_USER="root:root"
See also issues
Issues
For sip clients to call in we had have to run Asterisk as root, this needs to be fixed. Asterisk on alpine runs as root out of the box.
Memory-usage
$ free
total used free shared buffers Mem: 185824 51772 34052 0 184 Swap: 0 0 0 Total: 185824 51772 134052
Call logging
To log phonecalls to a csv-file, edit : /etc/asterisk/modules.conf
$ vi /etc/asterisk/modules.conf
And add load => cdr_csv.so:
; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=no load => cdr_csv.so etc
Now your logs will be in /var/log/asterisk/cdr-csv/Master.csv
For other forms of logging, see [here]
This box is running the following services: Ssh, asterisk, tinc and openvpn All this would fit in 64mb.
Resources
A very nice series of articles called "VoIPowering Your Office with Asterisk: SOHO VoIP" can be found below:
- Intro, How to connect an Asterisk server to legacy phones and phone service, [Part 1]
- Set up a connection to the outside world and set up internal extensions, [Part 2]
- Configure outbound calling, [Part 3]
- Voicemail, [Part 4]