Summary:	ASTERISK-16155: Speakerphone LED Update Fail after call termination.
Reporter:	Tom Gilheany (tom_gilheany)	Labels:
Date Opened:	2010-05-26 18:26:09	Date Closed:	2011-06-07 14:05:03
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_unistim
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
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). ****** ADDITIONAL INFORMATION ****** Steps to induce the problem (problem may happen in other circumstances as well). (0) Leave handset in cradle (on-hook position). (1) Dial an extension number using the keypad. Example [1][0][1] (2) Press the softkey1, labled as [Call]. (3) LED indicator for speakerphone lights. (4) Call Audio begins (ringing on speakerphone). (5) Far side answers (6) Far side terminates the call. (7) Speakerphone LED remains lit, despite call termination. Workaround: (1) Pick up handset on affected phone. (Offhook, listen to dialtone). (2) Replace handset on affected phone (On Hook). (3) Speakerphone LED turns off.
Comments:	By: Alec Davis (alecdavis) 2010-05-26 23:45:30 Sounds like an option on the phone. On a Grandstream GXP2000 it's <b><u>Turn off speaker on remote disconnect</u></b> By: Tom Gilheany (tom_gilheany) 2010-05-27 10:44:04 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). By: Paul Belanger (pabelanger) 2010-06-25 09:19:22 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt By: Paul Belanger (pabelanger) 2010-07-19 10:27:21 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines