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

[noreply at bugs.digium.com]

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:                
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Project:                    Asterisk
Issue ID:                   8824
Category:                   Core/General
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 59043 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             01-15-2007 18:18 CST
Last Modified:              10-15-2007 11:45 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. 
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Relationships       ID      Summary
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related to          0006643 [patch] Implement Called Party Identifi...
has duplicate       0008990 Transfer and Variables
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---------------------------------------------------------------------- 
 andrew - 10-15-07 11:45  
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Phones tested:
Cisco 7960 with SIP 8.8: all display updates work, execpt when on hold.
When the call is taken off hold the display updates.

Polycom 650 with SIP 2.2.0: all display updates work, even when on hold.
Grandstream GXP-2000 with SIP 1.1.4.25, no display updates at all.

Sipura/Linksys SPA-942 with SIP 5.1.15(a), the first update works when
calling out (eg. to set the called name on the display) but additional
updates during transfers do not work.

Linksys WIP300 with SIP 1.00.09: no display updates at all. 

Issue History 
Date Modified   Username       Field                    Change               
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10-15-07 11:45  andrew         Note Added: 0072015                          
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