I just tried to call in after creating an account. After the call connects, enter the show id: 22622# and your_PIN#
I dial in and am asked for the podcast id, I enter 22622# and am told that my passcode is not correct. I also tried just entering my passcode but got the same error message. What am I doing wrong? Thanks, Steve randulo wrote: > Hi folks, > > The August 3 edition of our Friday conference call and podcast kicks > off in just over and hour. I know the list isn't delivering properly > but if a few people get this it'll be better than none. > > We are going to be talking today about TDM inside and outside the box. > I own some antiiquated X100P FXO and a couple of TDM400p with the FXS > modules. This is how our company's litle pbx talks to two incoming > POTS lines and three regular phones connected to it. It also has a > long list of IAX and SIP providers connecting it to the rest of the > world. I am currently in the US so I use one of my 800 numbers to take > control of the asterisk box in Paris and make "local" calls in France > for a few pennies a minute. We also can send and receive SMS and of > course receive vmail via email. > > But enough about me. What are you doing about connecting? And more to > today's point, what ATA are you using to connect without opening the > box and installing hardware? > > Digium makes the IAXy, Sipura (whatever the name is today) has several > SIP models, Grand<cough>stream as well. What else is out there and > how well do they work? > > Join us: > > http://AsteriskUsersConference.org > > As Matt said somewhere, this conference is like a forum. It's a chance > for you to give back some of the valuable information and experience > into the community without writing a line of code. I've been using > asterisk for a few years and while I don't write code for it, I've > experimented a lot with lots of hardware and a long list of providers. > I've had time to learn a lot about the real world of all this stuff > and I'm willing to share what I know. How about you? > > > > On 7/29/07, randulo <[EMAIL PROTECTED]> wrote: > >> Hi, >> >> I am going to be on the road for the next few days and with the >> variable delay on the mailing list, I am posting this now, 4 days >> before the conference. If you haven't yet listened or participated, >> please consider doing it. We have a great kernel of people at all >> levels of expertise and ideas and questions can be kicked around >> immediately (well, there's a few milliseconds lag). >> >> This Friday we'll be talking about TDM solutions including ATA that do >> IAX and SIP without opening the box and installing a card. Your >> experience in this area would be appreciated. If you sell these >> solutions come over and "pimp" them. >> >> You can find us here: >> >> http://AsteriskUsersConference.org >> >> At this site there are three main conference pages, how to listen or >> participate, a player page for the archived recordings and a page with >> the extension for a SIP connection to the conference bridge. There are >> also two links to other pages, a related blog and AsteriskTV which >> will be getting more and better content and more formats due to the >> issue of Flash not being compatible with 64-bit systems. I'm working >> on this now and hope to have that done by mid September. If anyone >> knows how to convert mp3 to oog on a FreeBSD system, let me know. The >> video issues are going to be more complicated so if you have >> suggestions, please post them or email them to me. >> >> Thanks to the numerous people who have been supportive of these >> efforts including Mark Spencer and the guys at Digium. >> >> randy >> >> > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users