[asterisk-bugs] [Asterisk 0017406]: Speakerphone LED Update Fail after call termination.

Asterisk Bug Tracker noreply at bugs.digium.com
Thu May 27 10:44:04 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17406 
====================================================================== 
Reported By:                tom_gilheany
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17406
Category:                   Channels/chan_unistim
Reproducibility:            always
Severity:                   tweak
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.31 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-05-26 18:26 CDT
Last Modified:              2010-05-27 10:44 CDT
====================================================================== 
Summary:                    Speakerphone LED Update Fail after call termination.
Description: 
Issue: Speakerphone LED indicator remains lit after termination of a call.

Phone Models tested: i2004, i2007, 1140E.

Server Version Info:
- Asterisk 1.4.31 (yum-updated from AsteriskNOW)
- FreePBX 2.7.0.2 (yum-updated from AsteriskNOW)
- chan_unistim-1.0.0.6aq.tar.bz2 (downloaded from mlkj.net).

Excerpt from unistim.conf:
;-----------------------------------------------------------
; Kitchen Phone - Extension 15 - i2004
;-----------------------------------------------------------
[Kitchen]                          ; name of the device
device=000ae40c1d4e ; mac address of the phone
rtp_port=10010             ; RTP port used by the phone, default = 10000.
RTCP = rtp_port+1
rtp_method=1                ; If you don't have sound, you can try 1, 2 or
3, default = 0
status_method=0          ; If you don't see status text, try 1, default =
0
titledefault=Kitchen x15 ; default = "TimeZone (your time zone)". 12
characters max
....

(in case status_method is involved with the problem).
====================================================================== 

---------------------------------------------------------------------- 
 (0122565) tom_gilheany (reporter) - 2010-05-27 10:44
 https://issues.asterisk.org/view.php?id=17406#c122565 
---------------------------------------------------------------------- 
No, this is definitely an issue with the chan_unistim.
- Grandstream GXP2000 runs its own local SIP client, which has local
client control over all sorts of hardware settings, including LED state.
- On a phone running Unistim (similar to other "thin" protocols), there
are almost no local-client options on the phone setup; Everything is
directly handled by the call server.  The call server sends all messages to
the display, interprets buttons pressed, and tells the phone what tones to
play, or what audio stream to start.  Using thin clients on the phones
allows centralized control (a single change at the server doesn't mean
walking around and adjusting all of the clients, and the phone can support
a wide range of user-interfaces, depending on what it is plugged in to). 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-05-27 10:44 tom_gilheany   Note Added: 0122565                          
======================================================================




More information about the asterisk-bugs mailing list