Live Interactive VoIP Users Conference About Asterisk with Mark Spencer of Digium

The VOIP Users Conference has arranged a live interactive conference on January 4th, 2008, at 12:00 PM Eastern USA (11:00 AM CDT, 9:00 AM Pacific). The special feature of this session is that Mark Spencer, creator of Asterisk, a Linux-based open-sourced PBX and the founder, chairman and CTO of Digium, joins the VoIP Users Conference live for discussion and questions about asterisk and Digium.
The VoIP Users Conference is a weekly live interactive podcast about Voice over IP produced by Talkathon.org. The object of the Friday conferences is to discuss all aspects of VOIP, including applications, services, products and development.
Anyone can join the conference on a regular phone by following instructions given at http://VoipUsersConference.org. It is also possible to connect Asterisk directly to the conference server via SIP or dial in directly from a SIP client. Details are on the site. Listen anonymously using the player at http://food4wine.ning.com which is the Conference Community site. Talkshoe was discussed here.


Listen to recorded shows as there is a conference on every Friday.
listen to the live stream or recordings.

To call in to live conferences, you should sign up (free) at http://www.Talkshoe.com

How to configure your Asterisk to talk with talkshoe.

; In extensions.conf: define your PIN and the show id

[globals]
MY_PIN=0123456789 ; whatever 10-digit pin you register at Talkshoe.com

;;; You can now use 1# as a PIN
;;; However, this makes it hard for me to see who's there to call on you
;;; It would be great of people would create Talkshoe accounts and use names we can call on to speak.
;;; Either their IRC pseudos, company names or name/initial like
;;; Digium_guys, Steve_S, f_williams, Zeeek, russellb

CONFERENCE_CODE=22622 ; Voip users conference Talkshoe show ID

VOIP_USERS_CONFERENCE=1234 ; whatever extension you want to use to reach the conference

; Put the extension in a context, such as "talkshoe"
; (and make sure the context is included in a context you are going to dial from)

[talkshoe]
exten => ${VOIP_USERS_CONFERENCE},1,Dial(SIP/123@66.212.134.192,60,D(${CONFERENCE_CODE}#${MY_PIN}#))
Important: Recent Talkshoe conference hardware changes may make the DTMF not work.
Enter codes manually if you have problems.
 

tag: VoIP Conference, Asterisk, talkshoe, Talkathon.org, VoIP Users Conference, digium, Mark Spenser