[asterisk-biz] [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[Steve Totaro]

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 <spamsucks2005 at gmail.com> 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
>>
>>     
>
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