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

[Daniel - Asterisk]

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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