similar to: anyone have solution to dtmf problem in console driver?

Displaying 20 results from an estimated 50000 matches similar to: "anyone have solution to dtmf problem in console driver?"

2005 Aug 27
1
dtmf not being detected from viatalk
I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici
2013 Jul 05
1
problem with dtmf detection in asterisk 11
Hi. I am having problems with asterisk detection dtmf properly in asterisk 11. I am up to rev 390229. Now, when coming in off a did we have with Velocity, the dids work fine, but from extensions often it misses digits -- I can type *4 and it will miss the 4. Often, if I type quite slowly things will work properly. All dtmf modes are set to rfc2833. Strangely enough, I did not notice this with
2010 Jun 03
1
11.6.2 segfaults after dtmf on dahdi channel
Hi. I have been using asterisk-1.6.2 and if I update the version -- using svn -- to around May 19 or after, when I dial a digit on my fxs port which is on an X400p card, asterisk seg faults. If I go back before about this date, this problem does not occur. The dahdi version is svn 7445. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question
2009 Apr 13
3
duration of rfc2833 generated dtmf
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your
2009 Dec 13
1
Random DTMF tones generated from speech
Thank you, very interesting! As I understand the Digium card is used as a interrupt source for Asterisk? Is there a diagnostic tool available ? Anybody else experienced a simmialr problem? Thank you! HB > From: > covici at ccs.covici.com > Date: > Sat, 12 Dec 2009 19:04:23 -0500 > To: > Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at
2007 Jul 01
1
problems with dtmf using asterisk-1.4 rev r 6745
OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if someone types an asterisk from the other end of a call, I here it forever till the call hangs up. Looks like it starts the vldtmf, but never ends it from the logs. I am using Digium 400P rev I with one fxs and one fxo module. Any way around this one? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How
2006 Dec 22
1
problems using the 1.4 version of meetme
Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do
2015 Dec 29
1
permission problems trying to access subdirectories of a samba share
Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 15:44, covici at ccs.covici.com wrote: > > Rowland penny <rpenny at samba.org> wrote: > > > >> On 29/12/15 13:59, covici at ccs.covici.com wrote: > >>> Hi. I am having problems accessing subdirectories on a samba share. I > >>> am using windows 10 build 10586 and linux kernel
2008 Feb 25
2
cannot dial out with latest zaptel and kernel 2.6.24
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack [Feb 25
2011 Jan 15
1
Problem with chan_dahdi and conferencing
Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load => chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) Now chan_dahdi is indeed
2019 Jan 24
2
trying to upgrade asterisk and Debian -- not working (John Covici)
What procedure did you follow to revert back to the old version? It sounds like your binary has been revereted, but the modules it needs to load are still the 13.24.0-rc1 modules... --- Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 which seems to be 13.24.0-rc1. At the same time I want to go from Debian 8 to DEbian 9 to get a more recent operating system and
2007 Mar 12
1
Problems with Voice conferencing
How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk(asteriskbunnies@yahoo.com) wrote > Hey! > > Thanks for your interest, i checked the modules and i
2015 Dec 29
2
permission problems trying to access subdirectories of a samba share
Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 13:59, covici at ccs.covici.com wrote: > > Hi. I am having problems accessing subdirectories on a samba share. I > > am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba > > 4.2.7. I have two shares, one called audio and the other called > > myshare. I cannot access the subdirectories
2009 Nov 01
1
asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension
Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no answer, the extension hangs up, but the dial status is busy instead of no answer. How do I get this to work -- do I need to update dahdi? The card is an X400p using its FXS module. Thanks in advance for any ideas on this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
2010 Sep 23
2
rtp problem with 1.8.0-rdc1
Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in
2012 Jan 01
2
asterisk 1.8 codec negotiation
Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by
2019 Oct 07
2
problem with new install with asterisk 15.7.4
Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph <gjoseph at digium.com> wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici <covici at ccs.covici.com> wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >> system and I am
2006 Feb 21
0
meetme feature request (or maybe its there already?)
Hi. There are two things in a meetme conference which I would like to be able to do -- one is to start recording somewhere in the conference without having to change my dialplan to use the r option -- sort of like the one-touch in the features.conf -- and some way to play some specifically named file fromwithin the conference. I have an extension which uses ControlPlayback to play a file, but
2011 Jan 07
1
system lockup when going into conference
Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with no Digium hardware -- and when I go into a meetme conference the system either locks up or is 100% cpu utilized or something -- I can't type anything and I have to physically reboot the system. The dahdi module is loaded and the last log entry is the playing of you are the only person in this conference,. How would I
2006 May 05
2
AW: AW: DTMF detection when outgoing call tomobilephones
Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there