[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 09:10 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 ...
====================================================================== 

---------------------------------------------------------------------- 
 (0091294) svnbot (reporter) - 2008-08-11 09:10
 http://bugs.digium.com/view.php?id=8824#c91294 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 137190

U   team/group/issue8824/apps/app_dial.c
U   team/group/issue8824/apps/app_queue.c
U   team/group/issue8824/channels/chan_agent.c
U   team/group/issue8824/channels/chan_dahdi.c
U   team/group/issue8824/channels/chan_h323.c
U   team/group/issue8824/channels/chan_iax2.c
U   team/group/issue8824/channels/chan_local.c
U   team/group/issue8824/channels/chan_mgcp.c
U   team/group/issue8824/channels/chan_phone.c
U   team/group/issue8824/channels/chan_sip.c
U   team/group/issue8824/channels/chan_skinny.c
U   team/group/issue8824/channels/chan_unistim.c
U   team/group/issue8824/include/asterisk/channel.h
U   team/group/issue8824/include/asterisk/frame.h
U   team/group/issue8824/main/channel.c
U   team/group/issue8824/main/dial.c
U   team/group/issue8824/main/features.c
U   team/group/issue8824/main/rtp.c

------------------------------------------------------------------------
r137190 | russell | 2008-08-11 09:10:25 -0500 (Mon, 11 Aug 2008) | 3 lines

Merge changes from the latest patch on issue
http://bugs.digium.com/view.php?id=8824, with changes to get it
to
apply to the latest code and compile properly.

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=137190 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-11 09:10 svnbot         Checkin                                      
2008-08-11 09:10 svnbot         Note Added: 0091294                          
======================================================================