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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-CA
style='font-size:10.0pt;font-family:Arial;color:navy'>I would recommend you to
call Unlimitel as they have a very good support. Or just send a copy of your
post to : <a href="mailto:support@unlimitel.ca">support@unlimitel.ca</a><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-CA
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-CA
style='font-size:10.0pt;font-family:Arial;color:navy'>David<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-CA
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=FR style='font-size:
10.0pt;font-family:Tahoma;font-weight:bold'>De :</span></font></b><font
size=2 face=Tahoma><span lang=FR style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>De la part de</span></b> Mike<br>
<b><span style='font-weight:bold'>Envoyé :</span></b> 7 septembre 2006
11:32<br>
<b><span style='font-weight:bold'>À :</span></b> 'Asterisk Users Mailing
List - Non-Commercial Discussion'<br>
<b><span style='font-weight:bold'>Objet :</span></b> [asterisk-users]
Asterisk hangs up after 10-15 minutes when SIPPhone is on mute </span></font><span
lang=FR><o:p></o:p></span></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi,</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I have a Polycom 501 connected to Asterisk 1.2.4 (and then
connected to a VOIP provider, Unlimitel in my case). My job requires me
to attend conference calls regularly, and I am usually there as a silent
listener. Therefore, I mute my phone.</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I`ve noticed that if I mute my phone, after 10-15 of being
muted, the line hangs up. I had the same problem with my GXP-2000 before,
so I dismissed the phone as being the problem. If I unmute regularly (or
the entire time), the line doesnt hang up (until it reaches max timeout of
course, which is much more than 15 minutes). So the problem is my phone
is muted. I have observed that about 6 times (out of 6 tries) in the last
4 months. It`s a reccuring issue for sure.</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>What I am left with is Asterisk (or my VoIP provider) as the
issue. Since I only have control on my own Asterisk server, I thought I
should start there. What setting could cause this? I have a fairly fancy
dialplan, but I havent changed anything else than the diaplan. All
system-wide Asterisk settings are default as far as I know.</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks,</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Mike</span></font><o:p></o:p></p>
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