[asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=8824 
====================================================================== 
Reported By:                gareth
Assigned To:                oej
====================================================================== 
Project:                    Asterisk
Issue ID:                   8824
Category:                   Channels/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0-beta9 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 59043 
Disclaimer on File?:        Yes 
Request Review:              
====================================================================== 
Date Submitted:             2007-01-15 18:18 CST
Last Modified:              2008-08-11 08:06 CDT
====================================================================== 
Summary:                    [patch] Remote (called) Party Identification -
chan_sip & chan_skinny implementation
Description: 
Overview:

This patch provides the ability to rewrite the called party information
on
channel types that support it.  Implementations for the SIP (see note
http://bugs.digium.com/view.php?id=1)
and Skinny (see note http://bugs.digium.com/view.php?id=2) channels have been
provided.

Current features are:

1. Make changes whilst the call is progessing though the dial plan, ie:

   exten => s,1,RemoteParty("Voicemail" <123>)
   exten => s,n,Answer()
   exten => s,n,VoiceMailMain()

2. When using call pickup it will rewrite the caller information showing
the caller that was picked up.

3. When unparking a call it will show the caller*id of the parked call.

The ability to rewrite the calling party identification on semi-attended
transfer is planned but doesn't work yet.

Implementation:

Transmission of the remote party data is done using indications with a
new
subtype of AST_CONTROL_REMOTEPARTY, format of the data is:

  "name" <number>|presentation

Any channel specific code is kept in it's _indicate() handler. Once the
channel driver has received the indication it uses the method specific to
it; in the case of SIP it sends a 180/183 response if possible and with
Skinny it uses transmit_callinfo().

Note http://bugs.digium.com/view.php?id=1: The SIP implemenation is only able to
update the remote party
before the call has been answered as there is no re-invite support yet.

Note http://bugs.digium.com/view.php?id=2: I don't have any Skinny phones so no
testing has been done on
that part. 
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0006643 [patch] Implement Called Party Identifi...
has duplicate       0008990 Transfer and Variables
related to          0011036 Crush at unknown place
related to          0012511 transfer number of caller to callee whe...
related to          0012357 [patch] add called/connected/busy name ...
====================================================================== 

---------------------------------------------------------------------- 
 (0091291) russell (administrator) - 2008-08-11 08:06
 http://bugs.digium.com/view.php?id=8824#c91291 
---------------------------------------------------------------------- 
I have some good news from the Digium side of things.  We now have some
development resources to put toward helping getting this completed.  We
have a few things on our list that we're planning on doing, including:

 - adding support for this functionality to both ISDN channel drivers
 - changing storage of this information on ast_channel to use a datastore
 - updating _all_ call transfer implementations throughout the code base
to support this new functionality
 - ensuring that connected party updates get sent any time a bridge
occurs
 - other minor things, like reducing a bit of code duplication

I'm going to create a branch in the team/group area where we will be doing
this work.  We hope to get this merged in as soon as we can! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-11 08:06 russell        Note Added: 0091291                          
======================================================================