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

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Dec 17 05:54:55 CST 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:              12-17-2007 05:54 CST
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 savag3 - 12-17-07 05:54  
---------------------------------------------------------------------- 
The svn r92267 patch appears to have a bug in app_dial.c which causes a
segfault. It the temp channel variable name is incorrect. It should be tc
as tmp->chan is not set until after this code is executed. 

The following patch fixes it:

--- apps/app_dial.c     (working copy)
+++ apps/app_dial.c     (working copy)
@@ -1432,9 +1432,9 @@
                        ast_set_flag64(tmp, DIAL_NOCIDUPDATE);
                }
                if (ast_test_flag64(peerflags, OPT_FORCE_CALLERID))
-                       ast_set_dialcallerid(tmp->chan, cid_num, cid_name,
chan->cid.cid_pres);
+                       ast_set_dialcallerid(tc, cid_num, cid_name,
chan->cid.cid_pres);
                else
-                       ast_copy_callerid(&chan->cid,
&tmp->chan->dialcid);
+                       ast_copy_callerid(&chan->cid, &tc->dialcid);

                /* Copy language from incoming to outgoing */
                ast_string_field_set(tc, language, chan->language); 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-17-07 05:54  savag3         Note Added: 0075540                          
======================================================================




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