search for: clearglobalvar

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no a...

...he will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is forwarded the call is still ringing? My current /etc/asterisk/extensions.conf file looks like this: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [pstn] exten => s,1,NoOp(Caller ID is ${CALLERID(num)}) exten => s,2,Dial(Zap/1,15,g2) exten => s,n,Congestion [local] ignorepat => 9 exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) exten => _9.,n,Congestion exten => 11,1,Dial(Zap/1,20,rt) Thank you in advance. -------------- nex...

I have a question about global variables. Is it possible to somehow keep global variables unset via Dial Plan even Restarting asterisk? tks Eduardo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130710/4df721b7/attachment.htm>

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1...

...t=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include => customer exten => h,1,Hangup exten => i,1,Congestion exten => i,2,Hangup [agnosco] include => local-extensions include => customer_ivr include => incoming [customer_ivr] include => local-extensions exten => s,1,Answer exten =...

...eople outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no ;OEM exten => _12125551212,1,Goto(OEM,s,1) [OEM] exten => s,1,Answer() exten => s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)}) exten => s,n,Background(Outsource) exten => s,n,WaitExten(10) exten => s,n,Goto(inside,133,1) exten => 9,1,Background(OEM_M...

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1...

...1] type=friend context=internal secret=6011 host=dynamic ;client on same LAN as Asterisk nat=no ;extension for IP phone [6012] type=friend context=internal secret=6012 host=dynamic ;client on same LAN as Asterisk nat=no ;====================== extensions.conf [general] static=yes writeprotect=yes clearglobalvars=no autofallthrough=yes [vosp-incoming] exten => s,1,Dial(SIP/6011) exten => s,n,Hangup [internal] exten => 6011,1,Dial(SIP/6011) exten => 6011,n,Hangup exten => 6012,1,Dial(SIP/6012) exten => 6012,n,Hangup include => outgoing [outgoing] ;Route calls starting with 0 to VOSP...

...m<NUMBER%7D at server1.domain.com> ,nopartial [55:44:33:22:11:00] model=symmetric host=server2.domain.com inkey=dundikey outkey=dundikey include=dundified permit=dundified qualify=yes order=primary ----------------------------------- extensions.conf [general] static = yes writeprotect = no clearglobalvars = no [globals] [default] include => internal include => parkedcalls [internal] include => external include => parkedcalls switch => DUNDi/dundified exten => 300,1,Dial(SIP/300) exten => 300,n,Hangup() exten => 5551234567,1,Goto(300,1) exten => 301,1,Dial(SIP/301) ex...

...060 <username> 8585 Registered It seems that all is all right. But, when I try to call a number from kphone (on datile3), I listen a message which say: the number you are dialing it does not exists. The .conf files: 1) extension.conf: [general] static=yes autofallthrough=yes clearglobalvars=no priorityjumping=no [eutelia] include => out_eutelia exten=>_XXXXXXXXXX,1,Dial(SIP/100@out_eutelia,20) exten => _XXXXXXXXXX,2,Hangup 2) sip.conf: [general] context=eutelia realm=voip.eutelia.it port=5060 bindaddr=0.0.0.0 srvlookup=yes...

...| username=ekiga1 | secret=6641ekiga1 | disallow=all | allow=ulaw | [ekiga2] | type=friend | host=dynamic | username=ekiga2 | secret=6641ekiga2 | disallow=all | allow=ulaw `---- And the significant lines of my extensions.conf ,---- | [general] | static=yes | writeprotect=no | autofallthrough=yes | clearglobalvars=no | priorityjumping=no | [globals] | CONSOLE=Console/dsp | IAXINFO=guest | TRUNK=Zap/g2 | TRUNKMSD=1 | exten => 555,1,Dial(SIP/ekiga1) | exten => 556,1,Dial(SIP/ekiga2) `---- Monroux Philippe (French Reunion island) -- - Pourquoi les capitalistes devraient installer Linux ? - Parce que c&...

Hello, I don't really understand how channel Local works. I need that asterisk initiate a call and get some data (DTMF). So to do that I've created this dialplan : ; extensions.conf - the Asterisk dial plan ; [general] static=yes writeprotect=no clearglobalvars=no [dtmf] exten => 1,1,Verbose(Get User ID) exten => 1,n,Dial(dahdi/1/99999999,120,G(read^1^1)) exten => 1,n,Hangup() [read] exten => 1,1,Hangup() exten => 1,n,Read(data,play-msg,4,,2,15) exten => 1,n,Verbose("${data}") exten => 1,n,AGI(send_data.py,${data...

...up=1 callerid=asreceived ;Sangoma A101 port 1 [slot:3 bus:2 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel => 1-15,17-31 Extensions ----------------------------------------------------------------------- [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [dundi-e164-local] switch => DUNDi/e164 [local] ignorepat => 9 include => default include => parkedcalls [d...

...G[3497]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' mi configuration files are this: extensions.conf: [general] static=yes writeprotect=yes ;autofallthrough=yes ;clearglobalvars=no ;priorityjumping=no [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =>_94XXXXXXX,1,Dia...

...r being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is supposed to be. I am thinking it is a problem between the zap interface and the PSTN. thanks extensions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] ATTENDANT=1001 OUTBOUNDTRUNK=ZAP/g1 [extentions] exten => _10XX,1,Ringing exten => _10XX,2,Dial(SIP/${EXTEN},20) exten => _10XX,3,Answer exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail) exten => _10XX,5,Hangup [voicemail] exten => _910XX,1,Wai...

...First problems: Why he don't see the extension ? sip.conf: [AS5300] host=192.168.50.125 context=as5300-incoming type=peer dtmf=rfc2833 nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp [as5300-incoming] exten => 0426000000,1,Ringing exten => 0426000000,2,Answer exten => 0426000000,3,Dial(SIP/Jpc,25,m) exten => 0426000000,4,Hangup And second problems: "Call from '...

...fromdomain=sip.externalpeer.com host=sip.externalpeer.com nat=never canreinvite=no [1111] type=friend username=1111 secret=<secret> host=dynamic context=tutorial nat=never insecure=invite qualify=yes -------------- next part -------------- [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp [tutorial] exten => _XXXXXXX.,1,Dial(SIP/${EXTEN}@provider,,r) -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk.logs.example1.txt.bz2 Type: application/x-bzip2 Size: 17606 bytes Desc: not available Url : http://l...

...aycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no ;define channels context=internal signalling=fxo_ks channel => 1-2 context=zapchans signalling=fxs_ks group=1 channel => 3 extensions.conf [general] static=yes writeprotect=no clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;TRUNK=Zap/g1 ; Trunk interface TRUNK=Zap/3 ; Trunk interface TRUNKMSD=1 ; MSD di...

...s) echotail=64 ;echo cancel tail setting devices=2 ;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) and the interesting lines from </etc/asterisk/extensions.conf>: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] PIERRE=Zap/1 MARC=SIP/marc PATRICK=Zap/3 PROSPECT=Zap/2 OPENSPACE=Zap/4 FT_FREE=Zap/5 FT_ALICE=Zap/6 VOIP_FREE=Zap/7 VOIP_ALICE=Zap/8 NUMERIS=CAPI/ISDN1 [macro-repondeur] ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device...

...erbuffer=no forcejitterbuffer=no autokill=yes calltokenoptional=192.168.0.20 [Srv2] type=peer host=192.168.0.20 qualify=yes trunk=no encryption=aes128 disallow=all allow=alaw allow=g729 context=Incoming peercontext=Incoming extension.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo [Incoming] exten => _X.,1,Playback(demo-thanks) exten => _X.,2,Hangup [Out] exten => _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r) exten => _201X...