Asterisk Users Conference


Introduction


The conference/podcast is a place to meet, exchange ideas, listen to experienced users, ask questions, get answers, and be an active member of the Asterisk community. The conference is meant to be open to all levels of asterisk expertise from total beginner to experienced developer. The idea is to share knowledge, to bring new people into the community and to promote this open software gem created by Mark Spencer. You can participate live during the conference or it's available in recorded form 24/7.

More info on members: The Social Network
Session recordings: Archive


Schedule


The Conference is Live Fridays at 12:00PM ET / 18:00 Paris Time

Format


The conference is a tool for the community and is meant to be relevant to all those who listen and participate. To that end, the format is usually very loose. There are often topics that are planned for the day but if the participants want to see another topic covered or a certain question posed then the conference will go in that direction.

There are certain conferences where a guest is scheduled to speak on a product or technology (see below for a list of previous guests). During these conferences the topic is a little more defined but there is most often time remaining to cover other topics.

In addition, much discussion happens before the show begins and after the show has ended. You may want to consider joining the conference a few minutes early and staying on after the recording has stopped.

To Participate


In order to listen/participate in the conference live you will need to sign-up for a free account at Talkshoe since you will need a PIN in order to access the conference. You can go here to sign-up with Talkshoe.
If you set your PIN to the callerid sent by your phone, the conference will be able to see who you are, making things easier for all.

PSTN


+1.724.444.7444
After the call connects, enter the show id: 22622# and your_PIN#
If you do not have a PIN, you can enter 1#
If your PIN is your callerID, enter 2# and callerID will be read



SIP


sip:7463#22622#PIN@proxy.ideasip.com (ulaw)
sip:200901@login.zipdx.com (g722)

sip:123@66.212.134.192
After the call connects, enter the show id: 22622# and your_PIN#
If you do not have a PIN, you can enter 1#


SIP from extensions.conf


If you add the following lines to your extensions.conf then you will have an extension within Asterisk that you can call in order to connect with the conference and you won't even have to remember the show ID or your PIN.

; 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 enter 1# if you do not have 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 ; asterisk users conference Talkshoe show ID

ASTERISK_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 => ${ASTERISK_USERS_CONFERENCE},1,Dial(SIP/123@66.212.134.192,60,D(${CONFERENCE_CODE}#{MY_PIN}#))


SIP header "hack"

Having trouble with DTMF? For years, Talkshoe was INBAND. Now it appears to be RFC, but for the last month or so, Talkshoe has added a method that should work better using the SIP Subject: header. The info you need looks like this:

Subject: <passcode>22622</passcode><pin>1234567890</pin>
Using the SipAddHeader application you can generate this header and DTMF should no longer be a problem.


Talkshoe Console


Talkshoe has a Java console application that you can use to listen to the conference as well as speak with others in the conference. The Talkshoe console application also has a text chat feature that can be used to chat with others who are also using the console application. One thing to be aware of if using the text chat feature is that there users on the conference who do not use the Talkshoe console and cannot see what you type in the chat window. You may want to consider using IRC (see below).

IRC


Since not everyone has the ability to run the Talkshoe console we also make use of IRC for text chat.

Server: irc.freenode.net
Channel: #voip-users-conference

If you are not familiar with IRC and need to download a client you can visit our list of IRC clients to get started.

Past Guests


  • Mark Spencer
  • Several Digium staffers
  • Allison Smith
  • Trixbox
  • Lumenvox
  • Nufone, Teliax, Junction Networks, VoicePulse
  • Adhearsion
  • Cognation
  • FWD
  • John Todd, Freenum.org

Archives


Even though all shows are archived it is certainly no replacement for being an active participant in the conference. While you can learn a great deal by listening to the shows in the archive you lose the benefits of actually being able to ask questions and participate actively in this valuable forum. We understand that you cannot be at every conference and that is why we have tried to make it as easy as possible to catch up on what you missed.

All shows are archived and can be downloaded or played using several different methods.


The Talkshoe archive.

The RSS Feed which is perfect for your media player such as iTunes.

Via a Flash MP3 Player.

and here Topics and downloads page.

Mailing List


There is a mailing list for the Conference on Google Groups.

The purpose of this group is to:
  • Provide notes of interest such as site URLs or names of products, companies or people to contact regarding a subject discussed
  • Solicit ideas and comments about what you would like see covered in future conferences
  • Announce subjects to be discussed in future conferences

You can subscribe to the mailing list or simply view the archive. There is also an RSS feed and an Atom feed.



See Also






Introduction


The conference/podcast is a place to meet, exchange ideas, listen to experienced users, ask questions, get answers, and be an active member of the Asterisk community. The conference is meant to be open to all levels of asterisk expertise from total beginner to experienced developer. The idea is to share knowledge, to bring new people into the community and to promote this open software gem created by Mark Spencer. You can participate live during the conference or it's available in recorded form 24/7.

More info on members: The Social Network
Session recordings: Archive


Schedule


The Conference is Live Fridays at 12:00PM ET / 18:00 Paris Time

Format


The conference is a tool for the community and is meant to be relevant to all those who listen and participate. To that end, the format is usually very loose. There are often topics that are planned for the day but if the participants want to see another topic covered or a certain question posed then the conference will go in that direction.

There are certain conferences where a guest is scheduled to speak on a product or technology (see below for a list of previous guests). During these conferences the topic is a little more defined but there is most often time remaining to cover other topics.

In addition, much discussion happens before the show begins and after the show has ended. You may want to consider joining the conference a few minutes early and staying on after the recording has stopped.

To Participate


In order to listen/participate in the conference live you will need to sign-up for a free account at Talkshoe since you will need a PIN in order to access the conference. You can go here to sign-up with Talkshoe.
If you set your PIN to the callerid sent by your phone, the conference will be able to see who you are, making things easier for all.

PSTN


+1.724.444.7444
After the call connects, enter the show id: 22622# and your_PIN#
If you do not have a PIN, you can enter 1#
If your PIN is your callerID, enter 2# and callerID will be read



SIP


sip:7463#22622#PIN@proxy.ideasip.com (ulaw)
sip:200901@login.zipdx.com (g722)

sip:123@66.212.134.192
After the call connects, enter the show id: 22622# and your_PIN#
If you do not have a PIN, you can enter 1#


SIP from extensions.conf


If you add the following lines to your extensions.conf then you will have an extension within Asterisk that you can call in order to connect with the conference and you won't even have to remember the show ID or your PIN.

; 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 enter 1# if you do not have 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 ; asterisk users conference Talkshoe show ID

ASTERISK_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 => ${ASTERISK_USERS_CONFERENCE},1,Dial(SIP/123@66.212.134.192,60,D(${CONFERENCE_CODE}#{MY_PIN}#))


SIP header "hack"

Having trouble with DTMF? For years, Talkshoe was INBAND. Now it appears to be RFC, but for the last month or so, Talkshoe has added a method that should work better using the SIP Subject: header. The info you need looks like this:

Subject: <passcode>22622</passcode><pin>1234567890</pin>
Using the SipAddHeader application you can generate this header and DTMF should no longer be a problem.


Talkshoe Console


Talkshoe has a Java console application that you can use to listen to the conference as well as speak with others in the conference. The Talkshoe console application also has a text chat feature that can be used to chat with others who are also using the console application. One thing to be aware of if using the text chat feature is that there users on the conference who do not use the Talkshoe console and cannot see what you type in the chat window. You may want to consider using IRC (see below).

IRC


Since not everyone has the ability to run the Talkshoe console we also make use of IRC for text chat.

Server: irc.freenode.net
Channel: #voip-users-conference

If you are not familiar with IRC and need to download a client you can visit our list of IRC clients to get started.

Past Guests


  • Mark Spencer
  • Several Digium staffers
  • Allison Smith
  • Trixbox
  • Lumenvox
  • Nufone, Teliax, Junction Networks, VoicePulse
  • Adhearsion
  • Cognation
  • FWD
  • John Todd, Freenum.org

Archives


Even though all shows are archived it is certainly no replacement for being an active participant in the conference. While you can learn a great deal by listening to the shows in the archive you lose the benefits of actually being able to ask questions and participate actively in this valuable forum. We understand that you cannot be at every conference and that is why we have tried to make it as easy as possible to catch up on what you missed.

All shows are archived and can be downloaded or played using several different methods.


The Talkshoe archive.

The RSS Feed which is perfect for your media player such as iTunes.

Via a Flash MP3 Player.

and here Topics and downloads page.

Mailing List


There is a mailing list for the Conference on Google Groups.

The purpose of this group is to:
  • Provide notes of interest such as site URLs or names of products, companies or people to contact regarding a subject discussed
  • Solicit ideas and comments about what you would like see covered in future conferences
  • Announce subjects to be discussed in future conferences

You can subscribe to the mailing list or simply view the archive. There is also an RSS feed and an Atom feed.



See Also





Created by: mmb74, Last modification: Sat 04 of Apr, 2009 (15:59 UTC) by randulo
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