[asterisk-users] OT - Callto:// tags options

[Mojo with Horan & Company, LLC]

Olivier wrote:
> Hello,
>
> I'm after something meaning "transfer ongoing call to the mentioned 
> phone extension" instead of "dial a new call".
>   
It may be that a PHP-based webpage that interacts with asterisk through 
the manager interface would be an easy way to accomplish this.

One would of course need to know the call-id of the call that matters, 
because a phone SIP/100 would make calls SIP/100-xxxxxxxx, for example.
However one found that out wouldn't matter.  Then, conceivably, a PHP 
file would be created that would take the SIP channel to redirect and the
new extension and context it should be redirected to:

http://pbx/redirect_call.php?chan=SIP/120-abcd1234&newexten=18005551212&context=outpstn

this file, redirect_call.php, would connect to asterisk via the manager 
interface and request the redirection using the variables from the HTTP 
GET request. 

So your webpage could use callto:// tags (or tel:// tags where 
appropriate, like polycom's microbrowser) to start new calls, and links 
with the above dynamically-
generated HREF for transfers. 

This was a great idea, Olivier.  Thank you for bringing it up.  I 
realized that this page could be made VERY effective by setting the 
initial URL for the
browser (in the case of the polycom microbrowser for example) to include 
the device requesting the page, for example 
http://server/index.php?me=SIP/120

That way, when one hit the Services button on a polycom with the 
microbrowser firmware, if they were NOT in a call, there could be a list 
of people to start a call to.
If they WERE in a call, the list of people would be people to whom they 
could transfer the current call to ;)  That's just too cool.

I'll post it to the list if I ever do it :)

Mojo