asterisk users - Oct 2008 - SIP channels seem not to close after call is finished

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*

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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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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> 
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-- 
Steve Murphy
Software Developer
Digium
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