[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:              04-21-2008 22:43 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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---------------------------------------------------------------------- 
 etom - 04-21-08 22:43  
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I tried the 1.4.19 patch on our system and still see some problems.  I'm
about to upload two debug logs that show the issues we have.

The first is a straight call from 8525 to 8581.  The call works fine, but
the RPID header sent to the caller is weird.  That happens on line 324 of
"directcall.log".  

The second is an aborted semiattended transfer.  8525 calls 8581, who does
an attended transfer to 8544, and completes it before 8544 answers.  After
that, 8525 ends the call, but still before 8544 answers.

You can see the same RPID weirdness on lines 291 and 905 for each of the
two initiated legs.  But also, the UPDATE method doesn't seem to work, and
after that the CANCEL results results in an Internal Server Error, and
repeated attempts of 8544 to answer the call get ignored.

The phones are Polycom 650s, SIP 3.0.1 , Bootrom 4.1.0
I've had to edit the log files to declutter them and eliminate the
messages from all the other phones on the system, so if there is a missing
(or extra) small message, that's why.  In sip.conf sendrpid = yes but I
removed the trustrpid lines I had earlier.

This is our production phone system, so I can only do limited testing, and
then only after business hours.  But I'd really like to get this
functionality in so let me know what I can do to help. 

Issue History 
Date Modified   Username       Field                    Change               
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04-21-08 22:43  etom           Note Added: 0085796                          
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