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Help / Asterisk

Help.Asterisk History

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September 10, 2007, at 12:42 PM by sve204 -
Changed lines 27-28 from:
http://forums.nyu.edu/cgi-bin/nyu.pl?enter=itp-telephony
to:
http://itp.nyu.edu/~sve204/telephony_list.html
Restore
April 05, 2007, at 05:44 PM by sve204 -
Changed lines 3-6 from:
We are in the process of migrating to a new Asterisk server (asterisk.itp.tsoa.nyu.edu) and upgrading Asterisk to version 1.4. Most of the information below is still relevant but there might be a couple of small inaccuracies. If you find any, please notify [[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]].

ITP has an Asterisk PBX setup asterisk.itp.tsoa.nyu.edu. The main phone number for this machine is 212.796.0729.
to:
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.
Changed lines 9-10 from:
To get started developing your own projects with Asterisk you need SIP based Voice Over IP account. There is no perfect (read: cheap and without limitation) service. Here are a couple of recommendations from your peers:
to:
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:
Changed lines 24-26 from:
You also need an understanding of Voice over IP and Asterisk. 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 [[http://www.oreilly.com/catalog/asterisk/|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 should also join:
to:
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 [[http://www.oreilly.com/catalog/asterisk/|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:
Changed lines 31-34 from:
Right now we are running a couple of projects off of the server. Here is information relevant to people with individual projects that have logins to the server.

The server is social so you can SSH or SFTP in using your net-id and password to asterisk.itp.tsoa.nyu.edu.
to:
Once you have an account you can SSH or SFTP in using your net-id and password to asterisk.itp.tsoa.nyu.edu.
Changed lines 38-41 from:
The first step in getting your SIP based VoIP service working with this Asterisk install is to go into the asterisk_conf directory and create three files:

The first 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.
to:
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.
Changed lines 44-45 from:
The next file should be named: 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).
to:
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).
Changed lines 65-66 from:
Last, you need to create a "dialplan". This should be yet another file in the asterisk_conf directory and should be the "context" for your project. Name this file: net-id_extensions.conf.
to:
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
Changed line 70 from:
[net-id]
to:
[NETID_default]
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February 10, 2007, at 05:14 PM by Shawn Van Every -
Changed lines 3-4 from:
We are in the process of migrating to a new Asterisk server (asterisk.itp.tsoa.nyu.edu) and upgrading Asterisk to version 1.4. Most of the information below is still relevant but there might be a couple of small inaccuracies. If you find any, please notify ([[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]]).
to:
We are in the process of migrating to a new Asterisk server (asterisk.itp.tsoa.nyu.edu) and upgrading Asterisk to version 1.4. Most of the information below is still relevant but there might be a couple of small inaccuracies. If you find any, please notify [[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]].
Changed lines 27-28 from:
If you are in need of an account on the server, you must talk with ([[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]]).
to:
If you are in need of an account on the server, you must talk with [[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]].
Changed lines 46-47 from:
The next file should be named: 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). You can look at js3646_sip.conf and sip_itp_broadvoice.conf for examples but here is the main structure:
to:
The next file should be named: 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).
Restore
February 10, 2007, at 05:12 PM by Shawn Van Every - Added link to new server wiki and redial notes
Changed lines 78-84 from:
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.
to:
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/
Restore
February 10, 2007, at 05:08 PM by Shawn Van Every - update to new asterisk server
Changed lines 3-4 from:
ITP has an Asterisk PBX setup on Social (social.itp.tsoa.nyu.edu). The main phone number for this machine is 212.796.0963.
to:
We are in the process of migrating to a new Asterisk server (asterisk.itp.tsoa.nyu.edu) and upgrading Asterisk to version 1.4. Most of the information below is still relevant but there might be a couple of small inaccuracies. If you find any, please notify ([[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]]).

ITP has an Asterisk PBX setup asterisk.itp.tsoa.nyu.edu. The main phone number for this machine is 212.796.0729.
Changed lines 27-28 from:
If you are in need of an account on the server, you must first talk with Nancy who can set you up with an account and then with ([[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]]) to get things going.
to:
If you are in need of an account on the server, you must talk with ([[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]]).
Changed lines 31-33 from:
The server is social so you can SSH or SFTP in using your net-id and password to social.itp.tsoa.nyu.edu.

In your home directory, you will see 3 symbolic links (directories):
to:
The server is social so 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:
Changed lines 40-41 from:
The first 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. You can look at js3646_sip_register.conf and sip_register_itp_broadvoice.conf for examples but here is a the main structure:
to:
The first 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.
Changed line 43 from:
register => phone#@sip.provider.com:secretword:phone#@sip.provider.com/phone#
to:
register => username:secretword@sip.provider.com/phone#
Changed line 58 from:
;context=net-id
to:
;context=net-id ; it is very important that any lines with "context" have the ";" in front of them. This comments them out.
Changed lines 67-68 from:
Last, you need to create a "dialplan". This should be yet another file in the asterisk_conf directory and should be the "context" for your project. Name this file: net-id_extensions.conf. Checkout js3646_extensions.conf and extensions_itp_broadvoice.conf for examples.
to:
Last, you need to create a "dialplan". This should be yet another file in the asterisk_conf directory and should be the "context" for your project. Name this file: net-id_extensions.conf.
Changed lines 73-78 from:
exten => s,1,NoOp,${CALLERIDNAME}
exten => s,2,Wait,2
exten => s,3,Answer()
exten => s,4,DigitTimeout,5
exten => s,5,ResponseTimeout,10
exten => s,6,Playback(invalid)
to:
exten => s,1,Wait,2
exten => s,2,Answer()
exten => s,3,Playback(invalid)
Changed line 78 from:
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://social.itp.tsoa.nyu.edu/asterisk/asterisk_reload.php Use the normal itp-student username and the normal password.
to:
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.
Restore
September 25, 2006, at 05:02 PM by Shawn Van Every - Formatting cleanup
Changed lines 11-15 from:
*Another interesting service is [http://www.sipphone.com/|SIPphone].
*Yet another interesting service is [http://connect.voicepulse.com/|VoicePulse Connect]
*I have heard good things about [http://axvoice.com|Axvoice] and [http://viatalk.com|Viatalk], both based in the New York area (-cory)
*ITP is using [http://www.junctionnetworks.com|Junction Networks] for service but there is a per minute charge.
to:
*Another interesting service is [[http://www.sipphone.com/|SIPphone]].
*Yet another interesting service is [[http://connect.voicepulse.com/|VoicePulse Connect]]
*I have heard good things about [[http://axvoice.com|Axvoice]] and [[http://viatalk.com|Viatalk]], both based in the New York area (-cory)
*ITP is using [[http://www.junctionnetworks.com|Junction Networks]] for service but there is a per minute charge.
Changed lines 18-21 from:
Along different lines: [http://www.lylix.net|Lylix.net] has a hosted Asterisk solution which may be necessary for some of your projects.

You also need an understanding of Voice over IP and Asterisk. 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 [http://www.oreilly.com/catalog/asterisk/|Asterisk: The Future of Telephony] also available as a download http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
to:
Along different lines: [[http://www.lylix.net|Lylix.net]] has a hosted Asterisk solution which may be necessary for some of your projects.

You also need an understanding of Voice over IP and Asterisk. 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 [[http://www.oreilly.com/catalog/asterisk/|Asterisk: The Future of Telephony]] also available as a download http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
Changed lines 25-26 from:
If you are in need of an account on the server, you must first talk with Nancy who can set you up with an account and then with ([mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]) to get things going.
to:
If you are in need of an account on the server, you must first talk with Nancy who can set you up with an account and then with ([[mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]]) to get things going.
Changed line 40 from:
[=
to:
[@
Changed lines 42-43 from:
=]
to:
@]
Changed line 46 from:
[=
to:
[@
Changed lines 61-62 from:
=]
to:
@]
Changed line 69 from:
[=
to:
[@
Changed lines 77-78 from:
=]
to:
@]
Restore
September 25, 2006, at 04:57 PM by Shawn Van Every - Formatting cleanup
Added line 8:
Changed lines 11-15 from:
*Another interesting service is SIPphone.
*Yet another interesting service is [VoicePulse Connect http://connect.voicepulse.com/]
*I have heard good things about [http://axvoice.com Axvoice] and [http://viatalk.com Viatalk], both based in the New York area (-cory)
*ITP is using [http://www.junctionnetworks.com Junction Networks] for service but there is a per minute charge.
to:
*Another interesting service is [http://www.sipphone.com/|SIPphone].
*Yet another interesting service is [http://connect.voicepulse.com/|VoicePulse Connect]
*I have heard good things about [http://axvoice.com|Axvoice] and [http://viatalk.com|Viatalk], both based in the New York area (-cory)
*ITP is using [http://www.junctionnetworks.com|Junction Networks] for service but there is a per minute charge.
Changed lines 18-21 from:
Along different lines: [http://www.lylix.net Lylix.net] has a hosted Asterisk solution which may be necessary for some of your projects.

You also need an understanding of Voice over IP and Asterisk. One of the best places to learn about these is at the Voip-Info Wiki: [http://www.voip-info.org/ http://www.voip-info.org/]. A good resource for Asterisk in general is [Asterisk: The Future of Telephony http://www.oreilly.com/catalog/asterisk/] also available as a download [http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11]
to:
Along different lines: [http://www.lylix.net|Lylix.net] has a hosted Asterisk solution which may be necessary for some of your projects.

You also need an understanding of Voice over IP and Asterisk. 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 [http://www.oreilly.com/catalog/asterisk/|Asterisk: The Future of Telephony] also available as a download http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
Changed lines 23-26 from:
[http://forums.nyu.edu/cgi-bin/nyu.pl?enter=itp-telephony http://forums.nyu.edu/cgi-bin/nyu.pl?enter=itp-telephony]

If you are in need of an account on the server, you must first talk with Nancy who can set you up with an account and then with ([mailto:Shawn.Van.Every@nyu.edu Shawn Van Every]) to get things going.
to:
http://forums.nyu.edu/cgi-bin/nyu.pl?enter=itp-telephony

If you are in need of an account on the server, you must first talk with Nancy who can set you up with an account and then with ([mailto:Shawn.Van.Every@nyu.edu|Shawn Van Every]) to get things going.
Changed lines 40-41 from:
<pre>register => phone#@sip.provider.com:secretword:phone#@sip.provider.com/phone#</pre>
to:
[=
register => phone#@sip.provider.com:secretword:phone#@sip.provider.com/phone#
=]
Changed line 46 from:
<pre>
to:
[=
Changed lines 61-62 from:
</pre>
to:
=]
Changed line 69 from:
<pre>
to:
[=
Changed lines 77-81 from:
</pre>

Once you have those files created, let Shawn know and he will activate your configuration in Asterisk and you should be ready to go.

After you have been activated with Asterisk you will need to "reload" if you make changes to these files. The current means to reload is via a script that I have setup on the machine: http://social.itp.tsoa.nyu.edu/asterisk/asterisk_reload.php I am password protecting this page as well. Use the normal itp-student username and the normal password.
to:
=]

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://social.itp.tsoa.nyu.edu/asterisk/asterisk_reload.php Use the normal itp-student username and the normal password.
Restore
September 15, 2006, at 11:56 AM by Shawn Van Every - Initial ITP Asterisk Server Information
Added lines 1-78:
'''ITP Asterisk Server Information'''

ITP has an Asterisk PBX setup on Social (social.itp.tsoa.nyu.edu). The main phone number for this machine is 212.796.0963.

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

To get started developing your own projects with Asterisk you need SIP based Voice Over IP 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 http://connect.voicepulse.com/]
*I have heard good things about [http://axvoice.com Axvoice] and [http://viatalk.com Viatalk], both based in the New York area (-cory)
*ITP is using [http://www.junctionnetworks.com 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: [http://www.lylix.net Lylix.net] has a hosted Asterisk solution which may be necessary for some of your projects.

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

ITP has a telephony listserv that you should also join:
[http://forums.nyu.edu/cgi-bin/nyu.pl?enter=itp-telephony http://forums.nyu.edu/cgi-bin/nyu.pl?enter=itp-telephony]

If you are in need of an account on the server, you must first talk with Nancy who can set you up with an account and then with ([mailto:Shawn.Van.Every@nyu.edu Shawn Van Every]) to get things going.

Right now we are running a couple of projects off of the server. Here is information relevant to people with individual projects that have logins to the server.

The server is social so you can SSH or SFTP in using your net-id and password to social.itp.tsoa.nyu.edu.

In your home directory, you will see 3 symbolic links (directories):
asterisk_agi, asterisk_conf and asterisk_sounds

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

The first step in getting your SIP based VoIP service working with this Asterisk install is to go into the asterisk_conf directory and create three files:

The first 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. You can look at js3646_sip_register.conf and sip_register_itp_broadvoice.conf for examples but here is a the main structure:

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

The next file should be named: 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). You can look at js3646_sip.conf and sip_itp_broadvoice.conf for examples but here is the main structure:

<pre>
[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
authname=phone#
dtmfmode=inband
dtmf=inband
canreinvite=yes
</pre>

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". This should be yet another file in the asterisk_conf directory and should be the "context" for your project. Name this file: net-id_extensions.conf. Checkout js3646_extensions.conf and extensions_itp_broadvoice.conf for examples.

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

<pre>
[net-id]
exten => s,1,NoOp,${CALLERIDNAME}
exten => s,2,Wait,2
exten => s,3,Answer()
exten => s,4,DigitTimeout,5
exten => s,5,ResponseTimeout,10
exten => s,6,Playback(invalid)
</pre>

Once you have those files created, let Shawn know and he will activate your configuration in Asterisk and you should be ready to go.

After you have been activated with Asterisk you will need to "reload" if you make changes to these files. The current means to reload is via a script that I have setup on the machine: http://social.itp.tsoa.nyu.edu/asterisk/asterisk_reload.php I am password protecting this page as well. Use the normal itp-student username and the normal password.
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Page last modified on September 10, 2007, at 12:42 PM