[asterisk-bugs] [Asterisk 0011814]: Hold button works in debug mode only

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11814 
====================================================================== 
Reported By:                tootai
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11814
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 99426 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             01-22-2008 04:51 CST
Last Modified:              01-23-2008 06:00 CST
====================================================================== 
Summary:                    Hold button works in debug mode only
Description: 
On our phone, the build-in Hold button is only working well when we put the
phone in debug mode with "sip set debug peer xxx". In normal mode, when we
press the button, other party has music on hold and then:

. if we wait more than 12 sec, our phone hangup the call, other party has
still music on hold and call will ended ONLY if party hangup

. if we want to take back the call under those 12 sec, the phone gets
silence, other party still has MOH. Hangup the call doesn't end it at other
party.

What we have in sip show channels after those tests:

Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format          
Hold     Last Message
192.168.10.7     DH-Poste1   7972b1e50d6  00102/15236  0x0 (nothing)   
Yes      Rx: INVITE
192.168.10.7     DH-Poste1   1aa733f9478  00102/56020  0x0 (nothing)   
Yes      Rx: INVITE
192.168.10.7     DH-Poste1   4eab2e1e4e8  00102/10642  0x0 (nothing)   
Yes      Rx: INVITE

Our setup (phones, server, dialplan) didn't change since ages and was
working well with a mid december SVN update. Seems that this problem
appears the same time that the one we had last week with G711U codec (ID
11777)

-- 
Daniel
====================================================================== 

---------------------------------------------------------------------- 
 oej - 01-23-08 06:00  
---------------------------------------------------------------------- 
Ok. As per the bug guidelines, I need you to turn on "core set debug" to 5
so I see what happens inside of asterisk. THanks. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-23-08 06:00  oej            Note Added: 0081071                          
======================================================================