[Asterisk-biz] Application bounty $200: app_sipredirect

[Race Vanderdecken]

Mere child's play for a good coder who has worked with SIP and Asterisk.

Lets see some more bounty money posted to help out JT.

Lets all work together to removed the Stigma of Asterisk be "Free"
software by using Capitalism as a market force for the good of all.

PS,
	Do you want the redirection tables in mysql or RADIUS?

Race "The Tyrant" Vanderdecken

How would you look up the  
-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of John Todd
Sent: Friday, January 21, 2005 1:28 PM
To: asterisk-biz at lists.digium.com
Subject: [Asterisk-biz] Application bounty $200: app_sipredirect


I would be willing to put up a $200 bounty for app_sipredirect if 
someone wants to write it.

Syntax:

   -= Info about application 'SIPRedirect' =-

[Synopsis]:
Sends a SIP 302 message to caller with custom content

[Description]:
SIPRedirect(extension[@host[:port]])
   extension := string which contains new extension (mandatory)
   host := hostname or IP address of SIP destination.  If left blank, 
this host's IP address will be used.
   port := integer.  If left unset, no port will be specified.


This application can only be called before a call is answered. 
Calling this application after a call has been answered, or if the 
originating channel is not a SIP channel creates a 0 result.  After 
being called successfully, the application exits with a -1 result. 
This seems to be opposite of what should happen, but there isn't much 
reason to continue processing the dialplan after handing off the 302 
redirect, and giving a "0" result would allow for graceful error 
handling and continued dialplan processing if non-SIP or previously 
answered calls were handed to the application.

The host information is tricky in the default situation: should we 
use the IP address that SIP is bound to?  Should we use the 
externipaddress value from sip.conf?  externhost?  Maybe.

The application would re-write the Contact: address on the reply, so 
that the newly formed URI would be used by the requester to 
re-initiate the call to a different location.

This nifty trick might be used to create a central "call diverter" 
which then redirects many end users to multiple endpoints without 
having to keep state.  The CDR would be very brief on the central 
host, and would not contain any data about activity that happened on 
the redirected host/gateway.  However, that's fine - this isn't 
designed to replace app_dial; it's just a crude method to distribute 
SIP callers when they might not really need to be attached to this 
particular Asterisk instance.

Anyone who wants to add to this bounty should reply to this post with 
their additional fund promise.  I will be liable only for my $200 - 
any other contributors will have to make arrangements separately with 
any potential authors.

JT
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