Displaying 20 results from an estimated 30000 matches similar to: "Listening for DTMF when dialling"
2005 Sep 27
0
Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)
Hi all,
I want to set up an extension which dials a group of phones while at
the same time plays a message ("Press 1 to leave a message") and
listens for DTMF. I haven't played around yet but the way I read the
docs this isn't possible as the dial command doesn't have appropriate
options and takes complete control of the channel. However surely this
is a normal thing to want
2005 Jul 25
1
Playing sounds while dialling
Hi all,
Does anyone know a way of playing a sound while the dial command is
running? I want to play a sound every 10 seconds while relevant phone(s)
are ringing and have ring tones played to the caller in the gaps.
I am aware of the fact that you can play a class of MoH during the dial
but I can see two problems with that.... firstly that the caller won't
hear the sound from the start and
2007 Jul 27
0
Keep playing Background while dialling invalid dtmf extensions
hi asterisk users
How can i make asterisk "ignore" invalid extensions, and go on playing the
background soundfile?
Normally, asteriks will take the user to the invalid extension if the caller
presses anything other than 1 or 2 in the following context::
[example]
exten => s,1,Answer()
exten => s,2,Background(hello-world)
exten => s,n,Goto(s,2)
exten =>
2007 Oct 24
1
Unusual DTMF behavior
We are having an issue where DTMF is not being sent out right away and the
tone duration is inconsistent. For a test we send a '5', then a second
later we send a '9', and then five seconds later we send a '5'. If you look
at the logs below you can see the first '5' is played right away, then the
'9' comes in and gets queued, but it doesn't start
2005 Sep 05
2
DTMF issue on IVR
Hi All,
I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with
RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt
and asks caller to dial four-digit extension. Caller has to dial slowly,
otherwise, Asterisk cannot recognize the extension number. I look at the
trace on Asterisk CLI and there are missing digit in the middle of string.
ex, caller
2009 Oct 07
1
DTMF Issues
I have a block of DID's that I ported to Vitelity about 7 days ago. The
problem is if a POTS caller dials into the system, his dtmf is not heard
at READ() or Background() while a prompt is played. After the prompt is
finished, then dtmf is heard. I've been working with their support, but
it still not resolved. SIP callers are not effected.
Yesterday, I purchased a DID from
2005 Jan 05
3
Sending DTMF to PSTN problem with SIP
Dear All ~
I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN & try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain seems to go wild (high), and the
DTMF tones are not recognized at the other end.
I tried setting the
2005 Oct 01
0
chan_zap vs. Panasonic DTMF integration
The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF
integration feature that can be enabled for dedicated voice mail ports.
What I want to do is connect an X100P FXO port to a jack on the
Panasonic and make use of the Panasonic's DTMF call progress tones that
it provides in DTMF integration mode.
The integration works well when a Panasonic extension is forwarding into
2015 Dec 11
0
Samba-4 DNS issue
Its should not be needed to drop and recreate.
Try the following.
Delete the created records in the "wrong" zone(s)
Try to delete the "wrong" zone. ( ignore the error)
Restart bind
Restart samba.
Pray and check again.
Are the "empty zones gone and are your dns records there again?
Greetz,
Louis
> -----Oorspronkelijk bericht-----
> Van: samba
2005 Jan 24
2
LiveVoip DTMF Issues
I have a couple of DID's with LiveVoip and am having major DTMF issues
on incoming calls. I am connecting to them through IAX using ULAW.
When someone dials one of these DD's (from a landline) they are for
the most part unable to navigate the IVR menu successfuly. I would say
the failure rate is greater than 80%. For example if the caller
presses 5 sometimes * will see the DTMF as 55 or
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 Feb 15
1
Symbian IAX client
Hi all,
Does anyone know of an IAX client for Symbian? I have an e61 and would
like to make calls through my home Asterisk box from places where I have
WiFi access, as NAT is in the way I suspect that it'll be a pain to get
SIP working like that as the NAT router doesn't do SIP connection
tracking.
Thanks,
Peter.
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2004 Aug 19
1
Isdn4Linux and DTMF
Hello all,
I currently have an Eicon Diva Client isdn card using i4l. Outbound
dtmf doesn't work (and never has), but there has been an annoying
problem with false dtmf detection in calls (that could be triggered
easily by blowing into the receiver on the remote end).
I looked through the list and found two patches that need to be
applied - 1 to isdn_tty.c in the kernel, and another to
2004 Jan 29
0
DTMF wrongly recognised
Before I start ... I know there's been a lot of talk over the last couple of
days about the list being slow etc... I understand there were problems and
that mail is starting to get through now. Just wondering - should I be
surprised if something I sent 48 hours ago hasn't turned up at all? And
something I sent about 6 hours ago still hasn't turned up (yet). I'm just
curious
2004 Apr 26
3
dtmf tone clamping in calls to external ivr
Hello,
I'm having trouble working out how to send DTMF tones to an external
IVR. My system has an analog phone connected to a TDM400P card, a SIP
software phone (Zultys LIPZ4) and is connected to a BRI in Australia
with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched
with the ISDN audio patch from Traverse (which allows the card to do
voice).
DTMF works fine between
2011 Jan 12
1
DTMF not being heard correctly by far end conference system
Hi there
I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some
2007 May 02
1
Reinvite after DTMF?
Is there a way to do the following scenario?
1) my asterisk box receives an incoming call from a toll free number
provider such as nufone, voicepulse, etc.
2) It then dials a number via SIP and outputs a DTMF sequence.
ok, that part we do every day.
3) After DTMF though, is it possible to get the two SIP channels
(original SIP caller plus SIP called) hooked together and have my pbx
no longer
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
2008 Mar 20
1
More DTMF issues
Still grasping at straws trying to solve DTMF detection issues with one
of my asterisk servers. This particular server is now running Asterisk
1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P
cards. I have tried adjusting channel gains, turning call progress and
relaxdtmf on and off, switching echo cancelers, just about everything
that Google turns up and I can't
2005 May 14
0
Transferring a call, IAX2->SIP, DTMF/RFC2833 doesn't work?
We are using Asterisk 1.0.7. We have this scenario:
IAX2 user comes in to Asterisk, dials an extension, and transfers to a
SIP user.
The dial command is simple, looks like this:
exten => 300,1,Dial(SIP/300)
Extension 300 is a legacy PBX device operated by touchtones. The user
(coming in over IAX2) is trying to drive this PBX using touchtones. But
the trouble is, by the time the touchtones