asterisk users - Apr 2005 - US$200 bounty for * paging feature

>> . close source and we own the code.
You are no better then Microsoft.

Hi All,

This is a * paging bounty. The requirements are,

. US$200 for this bounty when all done. Need paging by April 27 ($75 paid
after done and basic testing), web interface done by May 3 ($75 paid after
done and basic testing). $50 paid two weeks after when more extensive
testing is done and all issues fixed. If you feel the deadline is not
achievable, let me know and we can discuss.
. close source and we own the code.
. Work with the latest stable version, currently 1.0.7.
. An application or AGI script which supports zone paging,
. Multiple extensions can be added into one zone,
. When someone wants to page a zone, it dials an extension which invokes the
application or AGI, it should accept parameters such as
"<zonename>@<extension>". (We use polycom phones and
already have 'auto
answer' working.)
. user makes the announcement, for a small group, make it real time, for a
large group, record the message and play it for a group of users.
. An option to page X number of users in a zone at a time, this would be
useful if a zone is big. In this case, the system records the message and
pages X number of users in sequence,
. a zone is allowed to include one or more other zones. In this case, it
would include all users in the included zone. (make sure that there is a
logic to disallow recursive zone inclusion)
. When a zone is paged, system will get all users in the included zones.
Remove redundant users before doing the paging.
. User and zone information is stored in a backend mysql database. (db table
structure will be provided later)
. A web interface which create/delete a zone, create/delete a user from a
zone, move a user from one zone to another
. The application or AGI must be in either C or PHP. (sorry, no PERL or
other scripting language)
. The web interface must use PHP with mysql as a backend database
. last but not the least, a easily readable and maintainable code is
expected. If you can please send some sample code you wrote before, that
would be really appreciated.

Thanks,
Richard

Richard wrote:
>. close source and we own the code.
>  
>
I was interested until I saw this. Nice try, but no thanks. If you are
doing what I think you're doing, you're opening Asterisk up to a patent
fight. And that's not what we're about. You want to develop for
Asterisk? Do it in the spirit of the community.

Since paging works just fine in the current code using external speakers or
speakers on the handsets, someone should take the $200 as partial
compensation for the aggravation. 

William Boehlke
Signate

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On 4/19/05, Mike <ast@uc9.net> wrote:
> >> . close source and we own the code.
> You are no better then Microsoft.
Speaking of an over reaction

-Matt

I am already doing this with AGI, PERL, and PHP to set up the page groups. 
I will release the code as open source if people are interested.  I'm not 
the best PERL scripter in the world but it works.

----- Original Message ----- 
From: "Richard" <richard@o-matrix.org>
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>; "'Asterisk Developers Mailing
List'"
<asterisk-dev@lists.digium.com>
Sent: Tuesday, April 19, 2005 5:56 PM
Subject: [Asterisk-Users] US$200 bounty for * paging feature

> Hi All,
>
> This is a * paging bounty. The requirements are,
>
> . US$200 for this bounty when all done. Need paging by April 27 ($75 paid
> after done and basic testing), web interface done by May 3 ($75 paid after
> done and basic testing). $50 paid two weeks after when more extensive
> testing is done and all issues fixed. If you feel the deadline is not
> achievable, let me know and we can discuss.
> . close source and we own the code.
> . Work with the latest stable version, currently 1.0.7.
> . An application or AGI script which supports zone paging,
> . Multiple extensions can be added into one zone,
> . When someone wants to page a zone, it dials an extension which invokes 
> the
> application or AGI, it should accept parameters such as
> "<zonename>@<extension>". (We use polycom phones and
already have 'auto
> answer' working.)
> . user makes the announcement, for a small group, make it real time, for a
> large group, record the message and play it for a group of users.
> . An option to page X number of users in a zone at a time, this would be
> useful if a zone is big. In this case, the system records the message and
> pages X number of users in sequence,
> . a zone is allowed to include one or more other zones. In this case, it
> would include all users in the included zone. (make sure that there is a
> logic to disallow recursive zone inclusion)
> . When a zone is paged, system will get all users in the included zones.
> Remove redundant users before doing the paging.
> . User and zone information is stored in a backend mysql database. (db 
> table
> structure will be provided later)
> . A web interface which create/delete a zone, create/delete a user from a
> zone, move a user from one zone to another
> . The application or AGI must be in either C or PHP. (sorry, no PERL or
> other scripting language)
> . The web interface must use PHP with mysql as a backend database
> . last but not the least, a easily readable and maintainable code is
> expected. If you can please send some sample code you wrote before, that
> would be really appreciated.
>
> Thanks,
> Richard
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote:
> 
> Michael D Schelin wrote: 
> > Ok you guys enough.  The debate will go on forever.  
> Agreed!  At the risk of wasting bandwidth myself....
> 
> Please, guys.... stop wasting my precious bandwidth.  If you want to
> private message your flames, great.... but leave this list to
> Asterisk, please.
> 
> Thanks!
> - Dan
Interesting that so many people are coming out to say stop, even to
reply to others saying stop and holding precious bandwidth up as the
reason.  I love your logic.

To jump on the bandwagon stop waasting my bandwidth telling people to
stop wasting your bandwidth.  Its only fair.

-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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stop wasting my bandwidth plz

On Wed, 20 Apr 2005, trixter http://www.0xdecafbad.com wrote:
> On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote:
>>
>> Michael D Schelin wrote:
>>> Ok you guys enough.  The debate will go on forever.
>> Agreed!  At the risk of wasting bandwidth myself....
>>
>> Please, guys.... stop wasting my precious bandwidth.  If you want to
>> private message your flames, great.... but leave this list to
>> Asterisk, please.
>>
>> Thanks!
>> - Dan
>
> Interesting that so many people are coming out to say stop, even to
> reply to others saying stop and holding precious bandwidth up as the
> reason.  I love your logic.
>
> To jump on the bandwagon stop waasting my bandwidth telling people to
> stop wasting your bandwidth.  Its only fair.
>
> -- 
> Trixter http://www.0xdecafbad.com
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
>

I just wanted to make everyone aware that I cross-posted my original 
message to the Biz list.  You may want to check out the responses there, 
too.

It looks like the entire Asterisk slave server pool in my diagram 
(http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) can be 
replaced by a VoIP gateway.  So far that seems like a better solution 
from every perspective. 

Matthew Roth
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

Callum McGillivray wrote:
> We are currently planning a large solution for a similar type of 
> scenario, ours is an implementation for a hotel with 600+ rooms.  The 
> core differences being that we will have a greater flow of concurrent 
> incoming and outgoing calls.
>
> I am also greatly interested in hearing about hardware setups for this 
> type of project.
>
> How many other people out there have done LARGE implementations ? We 
> would definitely like to hear about them.
>
> Another key factor with our installation is that we are going to have 
> several hotels of a similar size with calls trunked back to an 
> exchange and handed off in VOIP to our carrier.
>
> Who lays claim to the biggest * installation ?  And how was it done / 
> what issues were encountered ?
>
> Cheers,
>
> Callum

Matt Roth wrote:
> I just wanted to make everyone aware that I cross-posted my original 
> message to the Biz list.  You may want to check out the responses 
> there, too.
>
> It looks like the entire Asterisk slave server pool in my diagram 
> (http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) can be 
> replaced by a VoIP gateway.  So far that seems like a better solution 
> from every perspective.
> Matthew Roth
>
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
>
Hey Matt,

What kind of VOIP gateway are you looking at implementing, tips would be 
appreciated.

Cheers,

Callum

Callum,

We are very early in the research phase of this project, so I only know 
the names of a few manufacturers and a couple of models that we are 
looking at.  They are:

Cisco (AS5400)
Lucent (TNTs)
AudioCodes (Mediant 2000)
Quintum

A thread on this list titled "VOIP Gateways & Asterisk" has just 
appeared that you may want to keep an eye on.

Matthew Roth
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian 

> Hey Matt,
>
> What kind of VOIP gateway are you looking at implementing, tips would 
> be appreciated.
>
> Cheers,
>
> Callum

A newer name to the US market Planet VIP-2100/2400

http://www.planet.com.tw/news/productnews/VIP-2100.htm

Single and Quad T1 gateways.

On Tue, 2005-04-26 at 13:03 -0400, Matt Roth wrote:
> Callum,
> 
> We are very early in the research phase of this project, so I only know 
> the names of a few manufacturers and a couple of models that we are 
> looking at.  They are:
> 
> Cisco (AS5400)
> Lucent (TNTs)
> AudioCodes (Mediant 2000)
> Quintum
> 
> A thread on this list titled "VOIP Gateways & Asterisk" has
just
> appeared that you may want to keep an eye on.
> 
> Matthew Roth
>
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
> 
> 
> > Hey Matt,
> >
> > What kind of VOIP gateway are you looking at implementing, tips would 
> > be appreciated.
> >
> > Cheers,
> >
> > Callum
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> 
> 
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