Displaying 20 results from an estimated 8000 matches similar to: "tone generation"
2007 Apr 24
6
Digium card sale
Good morning,
Pardon for this intrusion I just wanted to let everyone know about some of
the specials that I have going on at HYPERLINK
"http://www.astawerks.com"www.astawerks.com . From now until the end of
June I will have a huge unpublished sale on all Digium products. Prices are
way to low to list so I will have to be personally contacted. I also have a
permanent sale on all
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323 connections
I am using Asterisk (Debian unstable packages) with an OH323 connection to my
provider. Everything is working except for the generation of ringback tones
when I receive inbound calls from the PSTN. My provider tells me that we're
sending call progress indications and that because of this they're expecting
us to generate the ringback tone. Does anybody know how to configure this in
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2004 Jul 25
1
how do I play congestion tone when Zap channels are full?
I read the wiki and looked at the examples, but I'm
still having problems. I have a Digium 4 port card
with POTS lines plugged into all four ports. How do I
play the congestion tone the the caller when they try
and dial out but all the lines are in use?
should something like this work?
[dial-trunklocal]
; Local calls
ignorepat => 9
exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1})
exten
2003 Jul 22
2
enabling dtmf detection on zap channel?
Hi,
is there a way to enable dtmf detection on zap channels? I am trying to
pickup, play a ringtone and the dial out. I.e.
exten => s,1,Wait,1
exten => s,1,Answer
exten => s,2,Playtones(dial)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => _X,1,StopPlaytones
exten => _X,2,Dial,Zap/g8/BYEXTENSION|10
2005 Mar 13
2
sending a DTMF tone before hangup
OK here is a possible tricky one.
I have a rocom door entry system which connects to an FXS port on my TDM400P
card. When the door button is pressed it initiates an 's' extension which
dials a number of SIP extensions. When a SIP phone is picked up the user
can speak to the person at the door and press the 7 digit which sends at
DTMF tone to the rocom unit opening the door. All this
2007 Oct 11
9
Mask Initial Processing with Ring Back Tone
I need to process a number of lines of code in the dialplan before answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate this?
Thanks in Advance,
Vic
2014 Oct 31
1
PlayTones while in call
I?ve gotten PlayTones to work, however it stops playing the tones as soon as
the call is answered. I would like to use PlayTones during the call because
I want to have a tone/beep played in the background while call recording is
going on.
Anyone know a way to get PlayTones to work while call is in progress?
Alternatively, does anyone have a suggestion for playing the tone/beep for
recorded
2007 Apr 15
9
Loudspeaker
Hello List,
This is what I want to do:
When a call comes in I want to ring an extension that happens to be loud
speaker. The users can the press *8 to answer the call. Is there a
SIP device that I can connect to Asterisk as an extension that can
accomplish something like this?
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2006 Apr 04
2
Milliwatt Test Number List
Hello:
Does anyone know of a list of milliwatt test numbers for debugging echo?
Specifically I am looking for a milliwatt test number in Canada,
preferrably in a 416 or 905 NPA exchange....different carriers would
also be nice....ie. Bell Canada, GT, Sprint (Now Rogers Telecom)
I called Rogers NOC and asked them for the milliwatt test number....they
didn't even know what it was....so I
2014 Oct 30
1
PlayTones not working
I?m trying to use Playtones to have a tone played periodically throughout
phone calls. Unfortunately, I can?t seem to get PlayTones to work. I never
hear the audio tones.
Here is the output on the Asterisk console.
-- Executing [19525553312 at proxy-dial:2] PlayTones("SIP/testphone-00000032",
"1400/500,2000/5000") in new stack
[2014-10-30 14:28:31] WARNING[23154]:
2006 Feb 18
1
snom 360 incorrect US indications
Anyone noticed the snom 360 indications are incorrect for US zone?
menu->preferences->tone scheme->usa
indications.conf:
[general]
country=us
extensions.conf:
exten => 1111,1,Answer
exten => 1111,n,Playtones(dial)
exten => 1111,n,Wait(30)
exten => 2222,1,Busy
exten => 3333,1,Answer
exten => 3333,n,Playtones(busy)
exten => 3333,n,Wait(30)
hit speakerphone on the
2007 Jun 19
1
Play dial tone withou answer
Hi,
I'm looking fore a way to play a dial tone before our IVR platform
answered the phone line.
I want to use for the following reason:
When a caller calls our Voice Platform, the call will direct dial out to
a number.
I want to dial out before the inbound call is answered.
But now the inbound call here's nothing.
When the outdial call is picked the inbound call will here
2006 Mar 02
5
Milliwatt Analyzer available
Hi,
some days ago we discused here the need for an analyzer
for the 1000 Hz tone, as opposite application to Milliwatt.
Here it is: Mwanalyze
http://planinternet.net/download/voip/asterisk/app_mwanalyze.c
It performs a Fourier analysis for a fixed frequency
and tells the amplitude.
The frequency is not limited to 1000 Hz, but can be passed
as argument. The periode duration must be a mulitple
2007 Nov 10
2
Record() : How to get filename created with %d?
Hello
About Record(), ATFT 2nd Edition says that "if the filename
contains %d, these characters will be replaced with a number
incremented by one each time the file is recorded."
Problem is, the documentation doesn't explain how to refer to this
filename later in the dialplan :-/
In this particular example, I want to move the file to the web
server's /htdocs so users can
2009 May 27
1
Playtones Volume
I've researched my brains out on this, and can't find any answer. Is there
a way to adjust the level of the tones generated through the Playtones
command? I'm thinking that I may have been approaching this incorrectly by
targeting indications.conf since the tones are being called via the
Playtones application. My sense is that it's not possible due to the lack
of response from
2004 May 29
4
PlayTones problem
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Hi!
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten => 123,1,Answer
exten => 123,2,PlayTones(Busy)
exten => 123,3,Hangup
But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call
just hangs up immediately.
I get the following on the console:
--
2009 Mar 19
4
"The number you have called has been disconnected or is no longer in service"
This sort of message is usually preceded by some magic tones that
allow direct marketing application to immediately drop a call to a
dead phone number.
What is the proper terminology for the tones?
Where can I find information about how this is implemented?
--
Drew Einhorn
2019 Jan 31
2
Dailplan with playtones
Hello I use this dial paln:
[o2-in]
exten => o2,1,Answer
exten => o2,n,Playback(hello-world)
exten => o2,n,Ringing
exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt)
exten => o2,n,Playtones(425/1000,0/4000)
exten => o2,n,Wait(30)
exten => o2,n,Hangup()
All is fine. Hello world is Playback and I hear a ring tone.
If I remove the Playback hello-world. No ring
2019 Jan 31
2
Dailplan with playtones
With softphone I mean linphone csipsimple or whatever.
How should a dialplan lokks like?
On 31.01.19 11:26, Antony Stone wrote:
> On Thursday 31 January 2019 at 10:59:01, basti wrote:
>
>> Hello I use this dial paln:
>>
>> [o2-in]
>> exten => o2,1,Answer
>> exten => o2,n,Playback(hello-world)
>> exten => o2,n,Ringing
>> exten =>