[asterisk-bugs] [Asterisk 0008824]: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Apr 3 18:58:55 CDT 2009
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: mmichelson
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Project: Asterisk
Issue ID: 8824
Category: Channels/NewFeature
Reproducibility: N/A
Severity: feature
Priority: normal
Status: ready for review
Asterisk Version: 1.6.0-beta9
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 59043
Request Review:
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Date Submitted: 2007-01-15 18:18 CST
Last Modified: 2009-04-03 17:41 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.
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Relationships ID Summary
----------------------------------------------------------------------
related to 0006643 [patch] Implement Called Party Identifi...
has duplicate 0008990 Transfer and Variables
has duplicate 0014271 SIP Remote-Party-ID not fully parsed
related to 0011036 Crush at unknown place
related to 0012511 transfer number of caller to callee whe...
related to 0012357 [patch] add called/connected/busy name ...
related to 0013690 CallerID not sent to SIP stations in SLA
related to 0012902 Video RTP is not sended to originating ...
related to 0014068 [patch] COLP/CONP support in QSIG
======================================================================
----------------------------------------------------------------------
(0102723) svnbot (reporter) - 2009-04-03 17:41
http://bugs.digium.com/view.php?id=8824#c102723
----------------------------------------------------------------------
Repository: asterisk
Revision: 186525
_U trunk/
U trunk/CHANGES
U trunk/apps/app_dial.c
U trunk/apps/app_directed_pickup.c
U trunk/apps/app_queue.c
U trunk/channels/chan_agent.c
U trunk/channels/chan_dahdi.c
U trunk/channels/chan_h323.c
U trunk/channels/chan_iax2.c
U trunk/channels/chan_local.c
U trunk/channels/chan_mgcp.c
U trunk/channels/chan_misdn.c
U trunk/channels/chan_phone.c
U trunk/channels/chan_sip.c
U trunk/channels/chan_skinny.c
U trunk/channels/chan_unistim.c
U trunk/channels/misdn/chan_misdn_config.h
U trunk/channels/misdn/isdn_lib.c
U trunk/channels/misdn/isdn_lib.h
U trunk/channels/misdn/isdn_lib_intern.h
U trunk/channels/misdn/isdn_msg_parser.c
U trunk/channels/misdn_config.c
U trunk/configs/misdn.conf.sample
U trunk/configs/sip.conf.sample
U trunk/include/asterisk/callerid.h
U trunk/include/asterisk/channel.h
U trunk/include/asterisk/frame.h
_U trunk/include/asterisk/rtp_engine.h
_U trunk/include/asterisk/stun.h
U trunk/main/callerid.c
U trunk/main/channel.c
U trunk/main/dial.c
U trunk/main/features.c
_U trunk/main/rtp_engine.c
UU trunk/main/stun.c
UU trunk/res/res_rtp_asterisk.c
------------------------------------------------------------------------
r186525 | mmichelson | 2009-04-03 17:41:48 -0500 (Fri, 03 Apr 2009) | 22
lines
This commit introduces COLP/CONP and Redirecting party information into
Asterisk.
The channel drivers which have been most heavily tested with these
enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support
and will be
introduced in a later commit. chan_skinny has code added to it here, but
according
to user pj, the support on chan_skinny is not working as of now. This will
be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball
rolling and
providing the initial support for this work. Without his initial work on
this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department,
as well as a
customer site running thousands of calls every day. In addition, many many
many many bugtracker
users have tested this, too.
(closes issue http://bugs.digium.com/view.php?id=8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
------------------------------------------------------------------------
http://svn.digium.com/view/asterisk?view=rev&revision=186525
Issue History
Date Modified Username Field Change
======================================================================
2009-04-03 17:41 svnbot Note Added: 0102723
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