[Asterisk-Users] signate.com webcall

[Kanuri, Seshu (Company IT)]

Hi All!

I already have this as a 'product' developed by Nicolas Gudino of Flash
Operator Panel especially for me as a fully functional system. You can
see this at http://www.eezeephone.com under callback services. Though it
may not be working now due to some misuse in the past.

Unfortunately what I have is not open source and it uses PHP/Mysql/ASTCC
as the backend combination and integrates all of them to be able to
Allow calls of only Authenticated users in Astcc. If anyone is
interested to improve the product further and relinquish their rights to
such improvements, wite to me off the list.

It is a very simple concept. Though what I have gotten developed comes
with a wider list of options - different Sets of Authentication,
Multiple Call Legs etc.

1)Use Asterisk Trivial Call Control Protocol
2)Take input from user for Source and Destinations
3)Take username and password for Authentication
4)Check if the Caller is in ASTCC database and he has balance
5)Create a Call file matching the source and destinations entered by the
user in /tmp
6)Copy the call file to Asterisk Spool
7)Update the calltime in ASTCC

(oops! I just gave away the design! Well, Asterisk and ASTCC is open
source, what the heck!)

Signate's version supports one call-leg between their extension and the
user's number, using PHP for the interface and use their DID as the
Called_From extension. They do not have a version that support two
distant Legs and Signate's system does not have an ASTCC integrated
CallingCard driven system, as I see it from the interface as it does not
ask for any authentication

Seshu Kanuri

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of VOIP
Consultant
Sent: Wednesday, April 20, 2005 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] signate.com webcall


   There are a several people providing that service.  The first time
the user invokes the service (clicks on the web link), he will have to
download the corresponding sip (or other) phone component.  Here is
where it gets difficult because it would have to be either a public
domain component or a
home made one.   And also it has to be a pretty good short component so
that
it doesn't take more than 10 seconds to download.  It is quite a cool
concept, I implemented it with a h323 DLL a while ago, but the guys that
worked it for me were concerned with the licensing issues.

C. Savinovich


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Moody
Sent: Wednesday, April 20, 2005 8:12 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] signate.com webcall


Signate offers an interesting product they call 'webcall', which
basically contacts a client at a number they provide then connects that
person to a sales staff. Some potential for abuse but a nice idea for
support etc.

I know that it is possible to do (obviously) and well documented but has
anyone actually released an open product similar to signate's webcall or
even a basic web initiated call interface (ie for calling cards).

I wasn't able to track via google or the wiki any ongoing projects - is
anyone interested in working on something like this?

J
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