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Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the Asterisk server memory. I tried to shut down the channels with the command "channel request hangup Khomp_SMS/B0C2-0" (Khomp_SMS/B0C2-0 the channel is locked), but nothing h...

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it mak...

...Local/761394357078@d s@default:1 Down (None) Local/78600054@defau 78600054@default:1 Up MeetMe(8600054|q) Local/78600054@defau 8309@default:3 Up Wait(3600) SIP/primus-95db 8600051@default:1 Up MeetMe(8600051) Zap/pseudo-122590356 s@default:1 Rsrvd (None) SIP/agent7-44fa 8600055@default:1 Up MeetMe(8600055) SIP/primus-0a7c 8600053@default:1 Up MeetMe(8600053) SIP/primus-7c73 8600054@default:1 Up MeetMe(8600054) Local/78600052@defau 78600052@default:1 Up MeetMe(8600052|q) Local/78600052@defa...

...system was working fine in the past. The motherboard was exchanged and during the switch, the phones line was rewired with a mistake and a incoming phone line was connected to the FXS module, and there was ringing voltage on the module. Now, the system kinda works but the FXS module stays in state Rsrvd, no ringing, all the call goes directly to voice-mail (it's programmed this way in extensions.conf), no dial-tone on the phone. Is my FXS module dead ? Thanks for any information Additional info: Asus A7A266 motherboard Gentoo with 2.6.9 kernel The system reboot daily at 4:52AM. It was upda...

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

...ion: 100 Priority: 1 Application: Playback AppData: vm-dialout Uniqueid: 1106323559.4 Event: Newexten Channel: IAX2/teenlighthouse/16384 Context: askdaqe Extension: 100 Priority: 2 Application: Dial AppData: Zap/g2/14356355785|10|tT Uniqueid: 1106323559.4 Event: Newchannel Channel: Zap/1-1 State: Rsrvd Callerid: <unknown Uniqueid: 1106323562.5 Event: Newstate Channel: Zap/1-1 State: Dialing Callerid: DAQE Dialer Uniqueid: 1106323562.5 Event: Newstate Channel: Zap/1-1 State: Ringing Callerid: DAQE Dialer Uniqueid: 1106323562.5 Event: Newstate Channel: Zap/1-1 State: Up Callerid: DAQE Dialer...

...p(DialStatus is ${DIALSTATUS}) exten => start,n,NoOp(${CHANNEL(state)}) exten => start,n,Playback(tt-monkeysintro) exten => start,n,Wait(2) exten => start,n,Hangup ============ ;tried this to force callfile to wait until off-hook [callbacktest] exten => start,1,Set(INDEX=0) ;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing ;Offhook, Pre-ring, Unknown ;Tried "Up", "OffHook", and "Rsrvd", to no avail exten => start,n,While($["${CHANNEL(state)}" != "Up" & ${INDEX} < 10]) exten => start,n,NoOp(Channel stil...

Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri -------------- next part -------------- An HTML attachment was scrubbed... URL:

...els when the conference is up in that configuration: asterisk*CLI> core show channels Channel Location State Application(Data) DAHDI/23-1 201 at Conferences:2 Up MeetMe(201,cosT) DAHDI/pseudo-1338070 s at default:1 Rsrvd (None) SIP/150-b444d988 201 at Conferences:2 Up MeetMe(201,cosT) What should I be looking at to debug this?

...answered? www.voip-info.org/wiki/view/Asterisk+func+channel www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS ============== [callback] exten => start,1,NoOp(DialStatus is ${DIALSTATUS}) ;how to pause until party has answered? ;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing Offhook, Pre-ring, Unknown exten => start,n,While([${CHANNEL(state) != "OffHook"]) exten => start,n,NoOp(Channel still ringing) exten => start,n,EndWhile() exten => start,n,Answer(500) exten => start,n,Playback(please-wai...

First of all - Many, many thanks to Mark for his troubleshooting and fix of bug 1320 (FXO_KS signalled Zap Channels on Adtran 750 Channel Bank Stuck in Rsrvd State). I have heard complaints that once every couple weeks, when a user picks up their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS cards), they don't get dialtone, but instead, hear another conversation. I'm under the impression that they can only hear and not sp...

...=> 'Newchannel', 'Uniqueid' => '1248244247.11', 'Privilege' => 'call,all', 'Channel' => 'Zap/1-1', 'CallerIDNum' => '<unknown>', 'State' => 'Rsrvd' }; ... EVENT: { 'Event' => 'Hangup', 'Uniqueid' => '1248244250.12', 'Privilege' => 'call,all', 'Channel' => 'Zap/2-1', 'Cause-txt' => 'Unknow...

In 1.4.43 I would see things from "core show channels" like DAHDI/18/xxxxx for line 18 in Asterisk 11 its DAHDI/i4/xxxx How do I get the line number back? Jerry

Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy Here is my /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=us defaultzone=us

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context