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

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Jun 18 13:29:37 CDT 2008


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:                   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:              
====================================================================== 
Date Submitted:             01-15-2007 18:18 CST
Last Modified:              06-18-2008 13:28 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
related to          0012511 transfer number of caller to callee whe...
====================================================================== 

---------------------------------------------------------------------- 
 francesco_r - 06-18-08 13:28  
---------------------------------------------------------------------- 
I have tested the lastest patch with a Linksys SPA941 (5.1.8) and a Snom
300 (6.5.13). The Linksys works with both RPID and P-Asserted. But still
doesn't work with Snom. Perhaps is a Snom problem because in the sip header
the p-asserted-field is regularly added. Following an ouput from a call
pickup:

<--- Transmitting (no NAT) to 192.168.1.58:2054 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.1.58:2054;branch=z9hG4bK-j8hmxbdv1esa;received=192.168.1.58;rport=2054
From: <sip:215 at 192.168.1.111>;tag=ldtgvxb5uk
To: <sip:*8 at 192.168.1.111;user=phone>;tag=as221c5d6d
Call-ID: 3c26720ac5c1-u6v8yxd87xq7 at snom300-0004132868B9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*8 at 192.168.1.111>
P-Asserted-Identity: "Interno" <sip:201 at 192.168.1.111>

It's possible to add in the patch the rewrite of CID when pickup from park
and with app_directed_pickup? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-18-08 13:28  francesco_r    Note Added: 0088875                          
======================================================================




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