[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:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   8824
Category:                   Channels/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0-beta9 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 59043 
Request Review:              
====================================================================== 
Date Submitted:             2007-01-15 18:18 CST
Last Modified:              2009-02-18 18:28 CST
====================================================================== 
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
has duplicate       0014271 SIP Remote-Party-ID not fully parsed
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 ...
related to          0013690 CallerID not sent to SIP stations in SLA
related to          0012902 Video RTP is not sended to originating ...
related to          0014068 [patch] COLP/CONP support in QSIG
====================================================================== 

---------------------------------------------------------------------- 
 (0100351) lacoursj (reporter) - 2009-02-18 18:28
 http://bugs.digium.com/view.php?id=8824#c100351 
---------------------------------------------------------------------- 
Hi, applied this patch to 1.4.23.1 cleanly.  Have Polycom 501, Polycom 500,
and Linksys PAP2T for testing.  All transfers worked fine, and directed
call pickup (**ext) works fine, but a general call pickup (*8) does not.

220 (Linksys) calls 223 (IP500)
222 (IP501) dials *8

IP501's display shows 220 (correct), and appears to be in a call, but the
Linksys is still ringing and there is no audio passed between them.

I actually tried this in all three possible combinations, and got the same
result :) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-18 18:28 lacoursj       Note Added: 0100351                          
======================================================================