similar to: Change time between DTMF

Displaying 20 results from an estimated 30000 matches similar to: "Change time between DTMF"

2014 Jun 07
2
asterisk-users Digest, Vol 119, Issue 7
I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To
2014 Jun 06
1
Shorten time between DTMF
I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal?
2007 Sep 14
2
DISA and DTMF detection problem w/ FXO port on a TDM400
-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using
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
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
2007 Jun 20
1
DTMF doesn't work between Asterisk and Cisco SIP Proxy
Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in
2004 Oct 05
1
difference between dtmf digit 8 and 9
Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting
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
2013 May 28
1
DTMF recognized after call establishment
Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:
2013 Jun 11
1
Why does it take several seconds to interpret DTMF-input ?
Hello, I notice that it takes 4 to 6 seconds between someone pressing a cipher and Asterisk continuing inside the dialplan. How come ??? Taken from verbose logfile : (attempt 1) [Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on SIP/SipAgenT01-00001eb0 [Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin ignored '1' on SIP/SipAgenT01-00001eb0 [Jun 11 15:29:25]
2008 Mar 10
2
About CID with DTMF in Asterisk
Hi, I have connected a TDM400P to my asterisk, I have enabled DTMF CID, the data is arriving to the asterisk but asterisk isn't interpretating it: its my full log: 1. Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0 2. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Starting simple switch on 'Zap/4-1' 3. [Mar 10 16:26:03] VERBOSE[9274]
2008 Dec 19
1
Increase DTMF Tone Duration
Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do
2011 Jan 05
2
DTMF-troubles with Snom
Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- <SIP/test1-00000701> Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan
2015 Feb 13
2
Debugging some DTMF Weirdness.
I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough
2017 Dec 13
2
DTMF emulation with SIP INFO and direct media
Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]
2009 Aug 25
0
DTMF duplicated when Waitexten
Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no
2009 Nov 20
1
How to change outgoing DTMF frequencies on zaptel?
Hi, I am having this issue that with one of the Asterisk servers, on zaptel hardware, that DTMF tones are 10-30 Hz too high than the upper limit for any DTMF digit frequency. This is causing problem with the equipment on the other end which is trying to recognize the DTMF digits. I tried different cards from the same vendor but it didn't help. For example for digit 1 where it should generate
2009 Sep 19
1
DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP
2007 May 03
1
Double DTMF digits
When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it
2008 Jul 07
1
DTMF on iax channel is not interpreted by asterisk
Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features