asterisk users - Sep 2006 - Asterisk hangs up after 10-15 minutes when SIP Phone is on mute

Hi,
 
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.
 
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.
 
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.
 
Thanks,
 
Mike
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In article <00ff01c6d292$c1d6dd20$0a01a8c0@MIKE>,
Mike <list@virtutel.ca> wrote:
>  
> 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.
>  
> 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.
Can you user Ethereal or similar to find out whether the phone stops sending
the RTP stream to Asterisk when it is muted?

Is there any kind of NAT gateway between the phone and Asterisk?

If both of the above are true, it might be possible that the NAT gateway's
session mapping for that stream is timing out.

Or perhaps Asterisk decides to hang up the call if it doesn't get any RTP
for a certain length of time.

Just a couple of ideas to try.

If the phone is still sending RTP when muted, the stream presumably contains
perfect silence. I don't know if that gets detected anywhere.

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

I would recommend you to call Unlimitel as they have a very good support. Or
just send a copy of your post to : support@unlimitel.ca

 

David

 

  _____  

De : asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] De la part de Mike
Envoy? : 7 septembre 2006 11:32
? : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone
is on mute 

 

Hi,

 

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.

 

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.

 

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.

 

Thanks,

 

Mike

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