search for: priorityjump

...[inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten => 441234123456,203,Dial(SIP/s1b,20,r) exten => 441234123456,304,Dial(SIP/s2a,20,r) i.e. try the first, if busy try the next etc. It seemed to consistently fail. in [globals] priorityjumping=no was set, which came from the samples (i.e. make samples when installing Asterisk). I changed that to yes (i.e. priorityjumping=yes) and it started to work. If that was the problem (which it seems to be), is that the wrong default? Or am I missing something here completely? Steve -- Net...

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SI...

...In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the "j" option in dial() application but no way. Here a sample of my simple dialplan : exten => 101,1,Ringing exten => 101,2,Answer() exten => 101,3,Dial(SIP/quentin,10) exten => 101,n,VoiceMail(101 at default,u) ex...

...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_Menu) exten =>...

I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag exten => 1234,2,Playback(digits/2) exten => 1234,3,Playback(digits/3) exten => 1234,102,Playback(digits/4) In this case, if I dial the extension, and it's answered, the caller...

Hi, I made a simple menu using the Background application and some wav files. I converted the wav files using for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done (from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk) The first two files "01/bemvindo" and "01/menu_top" are good.

...e> 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 defaulte...

...ecret=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'est le systeme...

...eived ;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 [default] exten =&g...

...__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,Dial(ZAP/g1/${EXTEN},...

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

...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,Wait(1) exten =>...

...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 '' to", AS53...

...fer=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 digits to strip (us...

...;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(s) to ring ; ex...

...tterbuffer=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.,2,Congestion...

...lbox=1001@voicemail callerid="jane doe" <1001> context=local-access nat=yes secret=password type=friend host=dynamic canreinvite=yes disallow=all allow=all 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 => _910...

...text=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid="my caller id"<(123) 123-1234> channel=>2 --------- /etc/asterisk/extensions.conf --------- [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password ;Changed from TRUNK=Zap/g2 ; Trunk interface TRUNK=Zap/1 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)...

Hi, although setting up voicemail for busy and unavailable should be easy, things aren't working the way they should in my configuration (asterisk 1.2.14 bristuffed): Here's the relevant part of the extensions.conf: exten => 56830976,1,Answer() exten => 56830976,2,Dial(SIP/hbaumgart,20,tr) exten => 56830976,3,VoiceMail,u76 exten => 56830976,4,Hangup exten =>

Thanks ... I forgot to say I tried it with priorityjumping=yes in the [globals] section of extensions.conf still no go... Gerald, I'll try your suggestion, and try to figure out the result code tests :-) Thanks, Rich > -----Original Message----- > From: Gerald A [mailto:geraldablists at gmail.com] > Sent: Tuesday, October 16, 2007 23:59...