Displaying 20 results from an estimated 42 matches for "priorityjumping".
2006 Mar 13
4
priorityjumping=no
...[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
--
NetTek...
2007 Jul 20
2
priorityjumping not working, Dial goes to n+1 not n+101
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("SIP/1...
2009 Jul 24
6
dialplan tips
...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)
exten...
2006 Nov 01
5
DTMF over IAX
...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 - ${CALLERID(number)})
exten => s,n,Background(Outsource)
exten => s,n,WaitExten(10)
exten => s,n,Goto(inside,133,1)
exten => 9,1,Background(OEM_Menu)
exten => 9,n...
2006 May 23
1
Problem with options to "Dial" application
I'm trying to set a dialing rule in my dialplan. As a part of it, from
my point of view, this works wrong
priorityjumping=no
[test_context]
exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag
exten => 1234,2,Playback(digits/2)
exten => 1234,3,Playback(digits/3)
exten => 1234,102,Playback(digits/4)
In this case, if I dial the extension, and it's answered, the caller ca...
2006 Feb 07
1
IVR Menu
Hi,
I made a simple menu using the Background application and some wav files. I converted the wav files using
for a in *.wav; do sox "$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.
2006 Oct 24
1
Basic Conf
...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
defaultexpi...
2007 Mar 10
1
installation pb on debian etch
...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 d...
2007 Aug 16
1
A102 card, BT ISDN30e, silence
...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 =>...
2007 Apr 27
1
can´t anserd the call
...__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},45,...
2007 Oct 14
4
ResponseTimeOut() and t extension
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?
2006 Mar 03
4
really need help with outgoing calls..PSTN errors
...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 => _91...
2009 Oct 28
2
Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
...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", AS5300...
2007 Jun 04
2
FX Dialing Odd
...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
(usual...
2006 Jun 19
2
Asterisk voicemail problem with isdn avm fritz!card
...;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
; ${ARG2} - Device(s) to ring
;
exten...
2009 Nov 21
3
Connect two Asterisk Server in IAX ?
...tterbuffer=no
autokill=yes
calltokenoptional=192.168.0.20
[Srv2]
type=peer
host=192.168.0.20
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontext=Incoming
extension.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[Incoming]
exten => _X.,1,Playback(demo-thanks)
exten => _X.,2,Hangup
[Out]
exten => _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r)
exten => _201X.,2,Congestion
==...
2006 Jan 31
3
ZAP <--> sip(polycom301) can not hear each other
...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 => _910XX,...
2006 Jun 16
9
Two FXO: How to dial a number when a RING comes in?
...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)
[...
2007 Mar 23
1
Problem with busy and unavailable
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 =>
2007 Oct 17
0
FW: DID to hunt group?
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
&g...