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: [EMAIL PROTECTED] 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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