Daniel - Asterisk
2008-Oct-14 22:24 UTC
[asterisk-users] 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081014/72edad82/attachment.htm
Steve Murphy
2008-Oct-15 20:29 UTC
[asterisk-users] SIP channels seem not to close after call is finished
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: 1.4.21.1These 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