[Asterisk-Dev] Tie VOIP into my web application

[Steven Critchfield]

On Thu, 2004-12-23 at 04:34 -0800, K J wrote:
> I want to tie my web application into a VOIP service so that users can
> call each other.  I want them to interface with my application's
> username system.
> 
> On the receiving user's end, he can either receive the call using a
> VOIP phone, or windows application (like skype).
> 
> I would use Skype's API, but it appears I need to use their username
> system, and not my own.
> 
> My question is, what software/hardware solutions would I need to do
> this?  Any suggestions/feedback would be greatly appreciated.

While this is a -user question, I'll give some answers so you are more
informed when you go to the -user list.

You can not mix real VoIP phones and Skype. Unless you have a lot of
bandwidth, you don't want to have your users calls coming through your
central server. You will want to use something like SIP all the way
through as it is currently the most likely protocol on a hard VoIP
phone. At that point, if you are just connecting users to users, you may
be better off with asterisk + ser. Asterisk can initiate the outbound
calls while ser handles all your users.

-- 
Steven Critchfield <critch at basesys.com>