[asterisk-users] SIP channels seem not to close after call is finished
Steve Murphy
murf at digium.com
Wed Oct 15 15:29:44 CDT 2008
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
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Steve Murphy
Software Developer
Digium
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3227 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20081015/8dbf94fb/attachment.bin
More information about the asterisk-users
mailing list