Asterisk

From SOWNWiki

Name	Asterisk
Deployed on	sown-dev
Website	http://www.asterisk.org/

Asterisk is an open source Voice over IP (VoIP) server.

It has been sugested that SOWN provide VoIP services to users of the SOWN network.

Implementation of SOWN VoIP has begun and details can be found on its own page.

This page provides further technical details about the use of the Asterisk VoIP server.

1 Current Setup
- 1.1 Database Setup
- 1.2 File Locations
2 Suggested Features for SOWN VoIP
3 Future work
- 3.1 University Phone System
4 Useful Scripts
5 Alternatives
6 Useful links

Current Setup

Asteriskv6 is running on sown-dev using realtime configuration (for sip devices and voicemail) from the asterisk mysql database running on the same machine

SOWN VOIP users can either be contacted using a user name or a corresponding phone number allocated to them at the sign up stage.

Database Setup

sip_conf: contains all data normally defined in sip.conf file, includes usernames and password hashes
voicemail_conf: contains mailbox definitions. mailbox names should correspond to usernames as stored in the sip_conf table
number_mapping: contains a mapping between numbers and usernames. This allows phones that cannot easily dial text, such as mobile phones, to call other users. It is also required for the ability to connect to other non voip phone systems
cdr: contains call records, including time of call, source, destination and call duration

File Locations

Dialplan setup: /etc/asterisk/extensions.conf
- Note: only text-based extensions should be stored in extensions.conf. All numeric mappings are done via the number_mapping table.
AGI Scripts directory: /var/lib/asterisk/agi-bin
Log file: /var/log/asterisk/
Web directory: /var/www/voip/

Suggested Features for SOWN VoIP

Please see SOWN VoIP for details of services that have actually been implemented.

SOWN Technical support number
Mobile IP testing
Free calls to university internal numbers
Voicemail
Paul enhanced talking clock
Peering with other university phone systems see http://www.sipphone.com/university/ucsd_peering.pdf

Future work

In future, we hope to interface with the university phone system.

This should be possible using a T1 line card from Digium [1]. I looked at it in the summer and it is definitly possible.

University Phone System

The university uses an Ericsson MD110 and talks to halls of residence and the hospital using DPNSS [2].

The system has several distributed PSTN exchanges. All the individual sites are interconnected with trunk connections.

Internally to Highfield campus, numbers are 5 digits starting with a 2. Halls phones begin with a 4.

The trunks between campuses add a prefix 7. So to call a line in Southampton general Hospital, you would dial 71xxxx where xxxx is the last 4 digits of the external phone number. To call Highfield Campus from SGH, you would dial 72xxxx. From Campus to Halls, you dial 78xxxx or 48xxxx depending on student or staff. New College is on 76xxxx.

External access is achieved by dialing 91 for University buisness or 92 for personal calls.

Dialling in from outside the exchange, one uses 023 8059 xxxx

Useful Scripts

see article: Asterisk Scripts

Alternatives

Are there any alternative VoIP/PBX systems that could be considered

Skype - Provides free calls to other Skype customers as well as regular phone lines world wide
Regular VoIP providers such as BT and Vonage - Similar to skype but more centeraly orientated, data goes through central servers, less P2P