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

noreply at bugs.digium.com noreply at bugs.digium.com
Sat Oct 27 10:16:59 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=8824 
====================================================================== 
Reported By:                gareth
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             01-15-2007 18:18 CST
Last Modified:              10-27-2007 10:16 CDT
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 lytledd - 10-27-07 10:16  
---------------------------------------------------------------------- 
New bug report for this patch.

Using the Asterisk feature "pickup extension" (mapped to *7), with this
patch enabled, will causes our Polycom IP501 phone to continue ringing and
act like it still has a inbound call; even though it was picked up by
someone in else in it's pickup group.  I get the console error:

[Oct 27 10:56:48] WARNING[25530]: chan_sip.c:12612 handle_response: Host
'10.10.10.176' does not implement 'UPDATE'

Disabling sendrpid for 4103 in sip.conf will fix the issue for that
phone.

As a review:

Asterisk 1.4.13
Polycom IP501, firmware 2.1.2
3 phones for testing, exten 4103, 4109, 4190
With sendrpid=yes on all 3 phones
4109 calls 4103, name shows correctly on both displays
4190 does a *7 to pick up 4103, call is picked up and display is correct
4103 continues to ring with correct callerid on the display

Picking up handset on 4103 makes no difference, phone still shows inbound
call
Pressing the Reject soft button will end the call.

Given 35 or so seconds, the phone will settle down on it's own and stop
ringing and clear the display.  I see the following error on the console
for 4103:

[Oct 27 10:56:48] WARNING[25530]: chan_sip.c:12612 handle_response: Host
'10.10.10.176' does not implement 'UPDATE' 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
10-27-07 10:16  lytledd        Note Added: 0072594                          
======================================================================




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