[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:                   Channels/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.0-beta4 
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:              03-19-2008 10:14 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
related to          0011036 Crush at unknown place
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---------------------------------------------------------------------- 
 yehavi - 03-19-08 10:14  
---------------------------------------------------------------------- 
Hello,
  The asterisk-1.4 patch worked for me correctly with version 1.4.18;
however, with version 1.4.18.1 it causes some crashes. I've fonud that
semi-attended transfers cause it to crash.
  I enclose asterisk_crash.log which is the output of Asterisk and
asterisk_core.log which is the log of the core dump.

  The logs are for a call from extension 80620 to 80623 (AudioCodes) which
answers it, transfer it to 85684 and then hangs up before 85684 answers (so
it is a semi-attended transfer).
                      Thanks, __Yehavi: 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
03-19-08 10:14  yehavi         Note Added: 0084240                          
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