ITP Help/F.A.Q. | Help / Asterisk

ITP has an Asterisk PBX setup asterisk.itp.tsoa.nyu.edu. The main phone number for student projects on this machine is 212.796.0729. An alternative number is 212.796.0961.

Yet another number for this machine is 212.796.0963 which leads to a project that is called the ITP Phone List and offers services for the ITP community.

Currently there are several experimental and student projects running on the server. It is also being used for the "Redial" class.

In the past, in order to get started developing your own projects with Asterisk you needed SIP based Voice Over IP account. This has now changed and when you request an account you will be given an extension. If your project is not public facing you are quite welcome to use this service but if you expect heavy usage or experience heavy usage you will need to purchase your own SIP based VoIP account.

There is no perfect (read: cheap and without limitation) service. Here are a couple of recommendations from your peers:

Broadvoice (Bring Your Own Device) is good and cheap but many uses of this fall outside of their terms of service which they don't like.
EXGN (http://www.exgn.net/) is the best per minute cost that I have found thus.
Another interesting service is SIPphone.
Yet another interesting service is VoicePulse Connect
I have heard good things about Axvoice and Viatalk, both based in the New York area (-cory)
ITP is using Junction Networks for service but there is a per minute charge.

There is a comparison engine here: http://voxilla.com/compare/compare.php?typeid=1

Along different lines: Lylix.net has a hosted Asterisk solution which may be necessary for some of your projects.

You will also need an understanding of Voice over IP and Asterisk in order to get started. One of the best places to learn about these is at the Voip-Info Wiki: [http://www.voip-info.org/]. A good resource for Asterisk in general is Asterisk: The Future of Telephony also available as a download http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

ITP has a telephony listserv that you MUST join: http://itp.nyu.edu/~sve204/telephony_list.html

If you are in need of an account on the server, you must talk with Shawn Van Every.

Once you have an account you can SSH or SFTP in using your net-id and password to asterisk.itp.tsoa.nyu.edu.

In your home directory, you will see 3 directories: asterisk_agi, asterisk_conf and asterisk_sounds

These are the 3 most important directories for the time being.

If you are setting up your own SIP service, you will need to create a "sip_register.conf" file. This file should be named: netid_sip_register.conf (replace netid with your net-id) and should have one line in it which is the registration information for your SIP VoIP service.

register => username:secretword@sip.provider.com/phone#

You will also need a: netid_sip.conf (again, replace netid with your net-id) and should have the information for peering your SIP based VoIP service with our Asterisk service (so you can make outgoing calls).

[net-id]
type=peer
user=phone
host=sip.provider.com
fromdomain=sip.provider.com
fromuser=phone#
secret=secretword
username=phone#
insecure=very
;context=net-id ; it is very important that any lines with "context" have the ";" in front of them.  This comments them out.
authname=phone#
dtmfmode=inband
dtmf=inband
canreinvite=yes

In the above two examples you will replace net-id with your net-id, sip.provider.com with the domain of your SIP provider, phone# with your phone number and secretword with your password. (Note: this is how it is setup with Broadvoice. If you are using another provider things might be a bit different.)

Last, you need to create a "dialplan" whether or not you have your own service. This file will live in the asterisk_conf directory and should be the "context" for your project. Name this file: net-id_extensions.conf. By default, your context should be named: NETID_default

Here is a very basic version which plays back an "invalid extension" message.

[NETID_default]
exten => s,1,Wait,2
exten => s,2,Answer()
exten => s,3,Playback(invalid)

After you have been created, you will need to "reload". You also need to do this if you make changes to these files. The current means to reload is via a script that we have setup on the machine: http://asterisk.itp.tsoa.nyu.edu/asterisk/asterisk_reload.php Use the normal "itp-student" username and the normal password.

For more information you are free to look over this wiki page (feel free to add to it): https://www.itp.nyu.edu/~sve204/cgi-bin/pwiki/wiki.pl?NewAsteriskServer

The Redial syllabus and notes are available here: http://www.itp.nyu.edu/~sve204/redial/

Page last modified on September 10, 2007, at 12:42 PM