Displaying 20 results from an estimated 50000 matches similar to: "WaitExten"
2009 Sep 02
2
DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
Is there any known reason that the DISA() routine should behave
differently than WaitExten() as far as recognizing DTMF tones? If
not, I suspect there's a bug here.
Try it yourself--two DID's on our PRI, numbers below let you test each routine:
It is my observation that some setups/phones DO and some DO NOT
express this variance.
--I could not show any variance on a sprint mobile phone
2011 Sep 18
2
DTMF problem
[This email is either empty or too large to be displayed at this time]
2004 Jun 18
3
WaitExten substitute
i am using the freebsd port, which seems to not yet have WaitExten(),
which i kinda want to use thusly
[ext-666]
exten => _.,1,SetVar(areacode=666)
exten => _.,2,Background(zz-in-who) ; give them list of extns
exten => _.,3,WaitExten(10) ; let them enter extn to call
include => extensions
include => applications
include => speeddials
2005 Sep 29
4
Calling voicemail from external phone.
Hey.
How would I set up my dialplan if a user wants to call its voicemail
from an external phone?
I'm thinking of getting the user to enter its mailbox number.
Something like this:
1. User calls the dedicated voicemail number.
2. Phone prompts for mailbox number.
3. Voicemail(${mailboxnr}@context)
Thanks.
2006 Jun 28
1
asterisk -> my cell phone's voicemail sound problems
When I fail to pick up a call from Asterisk to the PSTN to my cell
phone and let it go to voicemail, the sound quality is always really
bad. When I call my cell phone's voicemail a few minutes later, it's
really garbledy and sounds clipped or something.
I've tried using Monitor to record the sounds that are being played to
my cell's voicemail, and the monitored sound sounds fine
2008 Mar 12
3
DTMF problems while greeting is playing (Background())
Hi,
I have a Digium TE410p T1 card and I've noticed that under asterisk
1.4.17/18 I have problems detecting DTMF in IVRs. I think I've
narrowed the problem down to some sort of interference between the
greeting that is playing and the DTMF tones. DTMF detection seems to
work very reliably when I am in Read() or WaitExten(), but is
absolutely unusable while in Background().
I hope someone
2006 Nov 01
5
DTMF over IAX
Ok sorry for not being specific. I am having a problem when people
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
2007 Mar 15
1
asterisk n-way call problem
Hi,
i am using the n-way-call dialplan solution found on voip-info. i have
added its entry in applicationmap of features.conf file. the problem
is......its not working. to activate the n-way call i dial *0 but nothing
happens. i have played around with dtmf and codec settings but no success.
the extensions and sip configuration is below if you want to have a look. I
dont have any clue why its not
2004 Dec 21
1
Lets try this again then! Q: SIP error from dialplan I suspect!
I am playing with the dialplan to get it working and I have a challange
with this error. I can't find what it means on the wiki :(
Any sugestions would be helpful at being able to forward it to the SIP
phone if it is online and avaliable but then let that fail and drop into
voicemail if it is not online or is busy.
cheers
David
-- Executing Dial("IAX2/firefly@89280250/3",
2005 Mar 25
2
WaitExten question
I'm a bit confused about how WaitExten works. I assumed that when it
returns 0, the next priority in the extension would be executed, but
that doesn't seem to be the case. When I get to WaitExten and enter
extension 8, it plays the message, then Waits another 10 seconds and
times out.
[local]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten =>
2011 Mar 23
2
using ${EXTEN} with waitexten
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten =>
2007 Aug 02
1
A simple IVR extension problem
Hi list,
I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS
5.
I am having trouble to make my simple IVR extension work, here is relevant
config:
zapata.conf
----
context=incoming
signalling=fxs_ks
channel => 4
context=internal
signalling=fxo_ks
channel => 1
-----
extensions.conf:
----
[office]
exten => s,1,Dial(Zap/1,30)
[home]
exten =>
2016 Jun 30
3
how to join 2 channels using AGI/AMI
sorry for top-posting, the two topics started with 2 different reason
subject, but then we finished on the same problem.
btw,the 2 show channel are reported above:
the channel with DTMF working
kcenter*CLI> core show channel SIP/pbx2-000004b9
-- General --
Name: SIP/pbx2-000004b9
Type: SIP
UniqueID: 1467323106.1275
Caller ID: xxxx
Caller ID Name: xxxx
2007 Aug 03
5
Difference between WaitExten and TIMEOUT (response)
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
Regards
Bilal
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2004 Jun 02
1
DTMF and SIP
Hi
I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the
phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also
tried inband) and I get the following error:
june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
This means that I cannot get access to voicemail from the handsets
2008 Nov 18
1
setting up callback
Greetings Asterisk users!
I'm trying to setup Asterisk system to act as a callback system together
with callcentric (http://callcentric.com) but it appears that I hit common
DTMF issue and I want to workaround this problem. Basically my current
setup is the following:
1) I have dedicated Asterisk server that it is linked to my callcentric
account
2) I have US phone number (DID) from
2006 Dec 31
0
IAX & WaitExten
Hello list,
I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten.
To simplify the problem I've brought it down to the following scenario:
- 3 Asterisk Server A,B and C (central).
- A and B both register with C.
Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.
2016 May 09
3
Switching between Music on Hold streams. [13.8.2]
Hi there;
I didn't see any "G" option in the example above, and the usage for
the option parameters is entirely undocumented at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial
The G options are as below
G - If the call is answered, transfer the calling party to the
specified priority and the called party to the specified priority plus
one.
context
exten
2007 May 16
1
WaitExten not responding on key presses
Hi,
I have the problem that WaitExten is not responding to key presses. Here
are the sections from my extensions.conf:
[globals]
incoming_call=0
menu=0
announce=0
[internal]
exten => 777,1,Goto(hotline,${EXTEN},1)
[hotline]
exten => _X.,1,Set(CALLERID(name)=Hotline)
exten => _X.,n,Set(original_extension=${EXTEN})
exten => _X.,n,GotoIf($[${announce}=1]?4:10)
exten =>
2008 Jun 17
1
looking for help / input with Blind transfer from asterisk to zap
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel -> Asterisk -> SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension -> Asterisk -> Zaptel
During this whole process, the original channel off the trunk
(lineside T1) is