[asterisk-users] SIP channels seem not to close after call is finished

[Daniel - Asterisk]

I've restarted the service and zombie channels were killed.

Daniel

On Wed, Oct 15, 2008 at 3:29 PM, Steve Murphy <murf at digium.com> wrote:

> On Tue, 2008-10-14 at 17:24 -0500, Daniel - Asterisk wrote:
> > Hello everyone,
> >
> > I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of
> > my queue interfaces, despite the fact it is free at that time, can you
> > give help?
> >      1. I see many sip channels from that extension:
> > [root at mysweetpbx]# asterisk -rx "sip show channels" |grep 648
> >
> > Peer               User/ANR    Call ID                  Seq (Tx/Rx)
> > Format           Hold     Last Message
> > 192.168.25.29    648         7c24869b010  00102/00000  0x2 (gsm)
> > No       Tx: ACK
> > 192.168.25.29    648         26e8187a0a4  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         5289c52b77e  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         7a6243bc21e  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         32bcf3ea3f9  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         21ff7be5355  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         04725bda23e  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         2e9a9db559c  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         7fab5e8044d  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> > 192.168.25.29    648         11313fc173a  00102/00000  0x0 (nothing)
> > No       Tx: CANCEL
> >
> > 2. Asterisk version: 1.4.21.1
>
> These look a lot like the "Zombie Channel Bloating Death" problems
> we attacked over the last few weeks. Please see if the latest svn
> version
> of 1.4 has these problems still. In high-volume systems, this looked
> like
> a huge memory leak that would lead to death by swiftly using up memory,
> file descriptors, etc. until Asterisk ran out of virtual memory and
> crashed.
>
> There are a couple of code paths, one leaves CANCELED channels lying
> around, the other BYE'd channels.
>
> murf
>
> >
> > 3. I'm using SIP realtime peers, sip.conf configuration follows:
> >
> >
> > [general]
> > bindport=5060
> > bindaddr=0.0.0.0
> > context=default
> > language=es
> > rtcachefriends=yes
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> > rtpholdtimeout=300
> > rtptimeout=300
> > dtmfmode=rfc2833
> > videosupport=yes
> > progressinband=yes
> > allowsubscribe=yes
> > subscribecontext=extensiones
> > notifyringing=yes
> > notifyhold= yes
> > limitonpeers= yes
> >
> >
> > Daniel Arohuanca Lagos
> > +51 1 994149553
> > Lima-Peru
> >
> > _______________________________________________
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> --
> Steve Murphy
> Software Developer
> Digium
>
> _______________________________________________
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