Displaying 20 results from an estimated 7000 matches similar to: "Flash on X100P does not really flash."
2004 May 08
2
x100p / Answer-> Flash -> Dial
I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX
2006 Nov 08
2
flash transfer problem in asterisk integration with old PBX
I've tried to transfer a call using the Flash command, but with my
configuration it doesn't work.
I have a traditional PBX connected with a zap channel to Asterisk that acts
like an IVR:
TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk
>From the TELCO line I can make a call to the traditional PBX and reach
Asterisk, the IVR system on Asterisk answers the call and I can
2006 Jan 26
2
Transferring Using Flash
Greetings.
I am attempting to configure a system based on Asterisk 1.2.3 to be used
as a backup should our aging voice mail/auto attendant system fail, which
seems increasingly likely given its advanced years. The first part of this
task is getting the auto attendant feature to work correctly, which I
would have figured to be relatively easy. I have successfully built a menu
structure, but cannot
2004 Jul 07
1
Call files timeout on Flash command
I managed to sort out my earlier query regarding flash times (changed delay
in zapata.conf)
Now, I am getting a timeout after the Flash command in an outgoing call-file
based call:
-- Attempting call on Zap/1/108 for 567112@demo:4 (Retry 1)
> Channel Zap/1-1 was answered.
-- Executing Festival("Zap/1-1", "Dialling now") in new stack
== Parsing
2005 May 23
1
SendDTMF into a conference room
I have been trying to figure a way to SendDTMF into a MeetMe room using
the Manager API.
I can't redirect everyone into another context and then bring them back
because that would mess up my logic.
I am trying to use local channels and the originate Action to accomplish
this.
Exten: 3441115
Priority: 1
ActionID: actid-00000001
Context: senddtmftones
Action: Originate
Channel:
2004 Sep 01
1
X100P + Call-Waiting - Flash how-to.
Hi all
I'm pretty sure someone must have done this before but I couldnt find any
trace of it on the web so I thought I would drop a note about how I ended up
doing it. I have also posted this info on voip-info.
Warning : This is not very elegant and I'm currently trying to write a patch
in order to make it better but so far, this the only way I've gotten this to
work.
Scenario :
I
2006 Nov 27
0
flash transfer problem in asterisk with old PBX
Hi,
I've solved the flash transfer problem changing the flash time in the
zapata.conf file,
I've set:
flash = 200 (the defualt was 750 ms)
in the extensions.conf the code is for example:
exten => 42,1,Flash()
exten => 42,2,SendDTMF(42,250)
exten => 42,3,Hangup()
now the transfer with flash works correctly.
About the question whether my PBX expects a hook flash for
2006 Nov 03
1
SendDTMF() behaves strangely
Hi, everybody:
As part of a paging macro I'm using SendDTMF to send digits to the
called party.
The section looks like this:
exten => s,1,Wait(0.5)
exten => s,n,SendDTMF(9531290)
exten => s,n,Wait(1.0)
exten => s,n,Set(MACRO_RESULT=CONTINUE)
To test I direct the call to a live extension just to hear what's
happening -- what actually happens is that only the 9 is sent, and
2010 Feb 20
2
Sending a hook flash to a DAHDI channel
I've got a piece of CPE equipment that has an FXS port that I have tied
to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a
standard telephone connected to it, I get a dialtone. If I dial a digit,
and send a hookflash, the device will provide a dialtone back for the
next available channel on the device.
I'm trying to recreate this same behavior with Asterisk,
2010 Apr 30
2
Continuing after a TIMEOUT(absolute)
Greetings,
I'm trying to continue to do some processing after a TIMEOUT
(absolute). In my dialplan below, when a call comes in to [default],
I call macro-phonenum and pass it a timeout of 20 seconds. macro-
phonenum sets TIMEOUT(absolute), then loops saying the phone number
that was called (in MACRO_EXTEN). When the timeout expires I want to
call my macro-hangup (so it can say
2003 Dec 16
2
AT&T access code entry by Asterisk
I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following:
exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70)
There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the
2007 Jul 25
1
Post voicemail processing.
This 2 line code is doing what I wanted.
exten => 200,1,voicemail(200)
exten => 200,2,Hangup
What I've been told is that they want the 20 year old phone system to
light up the message bulb. (yea, a filament bulb, not an LED) To do
this you pick up on the line that goes into Asterisk and do a:
exten => 200,1,SendDTMF(200w#86)
But I don't know the path to take to get that
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
2017 May 23
2
Automatically dial a number, then an extension
On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote:
> All;
>
> What I did was add a line in the dialplan that used the SendDTMF()
> application and that worked great. The problem that I?ve run into now is
> that dialing the extension screwed up the answering machine detection. The
> sample context looks something like this.
>
> [play-audiomsg]
> exten =>
2005 Jul 11
2
DTMF not sending properly via IAX
I'm not sure if this is a -users or a -dev question, since the answer
probably comes down to something in the code.
I'm running the latest CVS-STABLE, and am subscribed to PSTN service
using IAX2 via Voiptalk in the UK.
I've just been alerted by a customer that the sending of DTMF from my
asterisk box to a remote PSTN user doesn't work, although it used to.
To test it, I have
2006 Nov 27
0
Queues and Flash/SendDTMF in hybrid PBX
Hi!
I am trying to setup a simple queue in Asterisk and
I'm having a small problem.
Our callers come in through a Bosch PBX and are
immediately transferred to an Asterisk menu/IVR. If
they select the option to call a SIP phone directly
(eg. entering the operator's SIP extension) then the
callee/operator can transfer the call to a phone
within the Bosch system. What Asterisk does is
2009 Nov 12
1
How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?
Hi,
After some testing I've found out that my client's hardware recognizes DTMF
only if digits are sent 50ms apart with 50ms of tone duration. This was
tested using a test device which generates DTMF.
Now asterisk doesn't do it by default because digits going out from Asterisk
are not being recognized.
Using command sendDTMF, I can control inter-digit duration, and using
2005 Sep 06
1
Some problems (SendDTMF, Wait, Parked Calls)
Hi all! I would like to solve some problems:
I have a sip provider that lets me make pstn calls after listening some
stuff and entering a pin number:
1) How can I make Asterisk enter the pin number? Then wait 1 second and
enter the phone number?
I have in extensions.conf:
exten => 6*,1,Dial,SIP/2002@myprovider,60,tr
I have tried with w (like with ZAP channels) but it does not work, nor
2003 Aug 05
4
SendDtmf
Hello all,
I am trying to use asterisk to call a local access gateway by dialing a fix number, after getting connected, the is a IVR prompt for pin number and finally the real destination number. I manage to use asterisk to dial to the gateway but have no idea how to send the pin number and destination number. This is due to asterisk only process the next ext only if dial app has terminated. My
2005 Jul 25
1
sendDTMF at pickup
Hi everyone:
The following code dials our prefix, sends a beep, and sends a DTMF "c"
tone, then dials the phone number.
I need to send the DTMF only if the phone is answered.
[voip]
exten=>i,1,NoCDR()
exten=>i,2,Hangup()
exten=>s,1,Wait(2)
exten=>s,2,Background(beep||)
exten=>s,3,DigitTimeout(6)
exten=>s,4,ResponseTimeout(10)
exten=>s,5,SendDTMF(c)