[Asterisk-Users] Zombie SIP channels

Pedro traci.asterisk at gmail.com
Thu Feb 10 06:26:13 MST 2005


Thanks for the feedback!

Running CVS-v1-0-11/12/04 (stable) on Fedora Core 1 with Cisco
7960G's.  Asterisk server is on public IP and Cisco 7960G is at client
location NAT-ed behind a Cisco soho91-k9 with nine other Cisco 7960G's
(each phone has registration expiring every 120 seconds).

Here is excerpt from sip.conf

[general]
disallow=all
allow=ulaw
port=5060  
context=incoming 
maxexpirey=3600
defaultexpirey=300
canreinvite=no
tos=reliability
srvlookup=yes
videosupport=no
dtmfmode=inband
nat=yes
insecure=very

[frontdesk]
context=customer
type=friend
username=frontdesk
secret=password
host=dynamic
canreinvite=no
mailbox=100 at customer
nat=yes
qualify=yes
callerid="Front Desk" <100>
accountcode=customer
amaflags=billing

This is the first time I have seen this so it does not appear to
happen too often.  Obviously would rather not upgrade if possible has
everything seems running fine.  But good to know that if it becomes a
problem, I can try upgrading to 1.0.3 or later.

Thanks!

Pedro


On Thu, 10 Feb 2005 09:19:45 +0100, Florian Overkamp
<florian at obsimref.com> wrote:
> Hi,
> 
> > -----Original Message-----
> > Does anyone know how to kill a zombie channel?
> >
> > Here is what I see on a show channels:
> > --------------------------------------------------
> > show channels
> >         Channel  (Context    Extension    Pri )   State Appl.
> > Data
> > SIP/frontdesk-72c7  (customercontext               1   )      Up
> > Bridged Call  SIP/frontdesk-0461<ZOMBIE>
> > SIP/frontdesk-0461<ZOMBIE>  (customercontext 100          1   )
> > Ring Dial          SIP/frontdesk|20|t
> > 2 active channel(s)
> > --------------------------------------------------
> >
> > No one is on a call - how can I get rid of this without
> > restarting asterisk?
> 
> This was an issue in older versions of asterisk. It would help if you could
> tell us what setup you are running.
> If this is infact your problem too, a simple update of your asterisk to
> 1.0.3 or later will help.
> 
> Florian
> 
>



More information about the asterisk-users mailing list