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

Tue Mar 17 15:20:28 CDT 2009

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=8824 
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Reported By:                gareth
Assigned To:                mmichelson
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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:              
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Date Submitted:             2007-01-15 18:18 CST
Last Modified:              2009-03-17 15:20 CDT
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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
====================================================================== 

---------------------------------------------------------------------- 
 (0101889) mmichelson (administrator) - 2009-03-17 15:20
 http://bugs.digium.com/view.php?id=8824#c101889 
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All right, everyone, this issue should hopefully be coming to a close very
soon. Asterisk 1.6.2 has been branched, meaning that Asterisk trunk is wide
open for new feature consideration. This would be a great candidate for
1.6.3.

For a look at the actual code which is being considered for merging into
Asterisk trunk, please see the branch located here:

http://svn.digium.com/svn/asterisk/team/group/issue8824

This branch is based off of the current trunk of Asterisk and contains all
remote party identification modifications. The branch started based off
gareth's excellent starting patches. Through testing conducted at Digium
and by others, we have modified it to accurately present remote party
identification through both normal calls and through features like call
pickup, transfers, etc. To help fast-track this, I have placed a diff
between the branch I linked above and trunk on review board, here:

http://reviewboard.digium.com/r/201/

While we have done extensive testing to be sure that we have implemented
the feature correctly, we still want to hear feedback if at all possible
before merging this into trunk. We expect that once the code is in trunk,
there will be more people who will test and report feedback.

My plan is to wait a couple of weeks to hear feedback both here and on
review board in case there are some major outstanding problems. Thanks for
your patience on this one. I know this is a much sought-after feature, and
hopefully it will be officially part of Asterisk very soon. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-17 15:20 mmichelson     Note Added: 0101889                          
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