[Asterisk-biz] Application bounty $200: app_sipredirect

[John Todd]

This patch has been completed, and is ready for testing.  The bounty 
has been fulfilled, and is no longer available.

http://bugs.digium.com/bug_view_page.php?bug_id=0003419

JT


At 10:27 AM -0800 on 1/21/05, John Todd wrote:
>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