[asterisk-bugs] [Asterisk 0016208]: Call Hold not working on some phones

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Nov 9 08:49:29 CST 2009


The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=16208 
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Reported By:                mustardman
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16208
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.0-rc4 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2009-11-08 15:54 CST
Last Modified:              2009-11-09 08:49 CST
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Summary:                    Call Hold not working on some phones
Description: 
Fresh install Asterisk 1.6.1.9 and addons.  Call hold works from Aastra 55i
with latest firmware and Xlite on SIP to SIP.

Fresh install Asterisk 1.6.2.0 RC4 and addons.  No changes to Aaastra or
Xlite config.  Call hold does not work on Aastra 55i.  CLI shows nothing as
if hold button is not being pressed.  Xlite hold button still works fine.
====================================================================== 

---------------------------------------------------------------------- 
 (0113410) lmadsen (administrator) - 2009-11-09 08:49
 https://issues.asterisk.org/view.php?id=16208#c113410 
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Due to the lack of information, I am closing this issue for now. When you
or someone else is able to provide additional information that would be
necessary to move this issue forward, then please reopen.

Because this is a SIP issue, the following information is required (quoted
from the bug-guidelines at
http://www.asterisk.org/developers/bug-guidelines)

"Include debug output! Please include output from "sip debug" if you have
a SIP problem. This seems obvious, but apparently is not. Set debug to 4,
verbose to 4, turn on sip history and dumphistory in sip.conf and capture
all output. A packet trace from ethereal will not tell us what is happening
inside your Asterisk server, so that is not a replacement." 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-09 08:49 lmadsen        Note Added: 0113410                          
2009-11-09 08:49 lmadsen        Status                   new => closed       
======================================================================




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