similar to: Random DTMF tones generated from speech in conversations

Displaying 20 results from an estimated 7000 matches similar to: "Random DTMF tones generated from speech in conversations"

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
2008 Oct 27
3
Door phone
Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway
2010 Jan 16
4
Howto regret blind transfer?
Hi, Is it possible to "regret" blind transfer while its ringing (not answered)? Thank you! Best regards HB
2004 Dec 06
3
Recomended ISDN for Asterisk ?
Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway
2009 Nov 11
2
Best practice to set up 4 line phones
Hi, I would like some advice from you on how to configure a multi line phone the best way! So far I have given the phone 4 sip accounts one for each line, this is a lot of work and gets messy. Is it a better way to do this? Thank you! Best regards Helge-Bj?rn
2005 Feb 07
1
Asterisk => SKYPE
Hi Any solution for connecting Asterisk to Skype without using fsx/fxo hardware ? Best Regards HB Norway
2009 Feb 11
2
DTMF tones mid conversation
Hi helpers, I seem to have a problem of intermittent DTMF tones being played during a conversation. Eg: Extn 100 takes an inbound call and all is fine. Except, at an undetermined time the person on extn 100 will here a DTMF tone for no apparent reason (it's not the caller pressing buttons). The caller doesn't hear the tone - only the called person. The call itself progresses normally.
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
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 Nov 06
7
DTMF Tones occuring randomly
Hi, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"
2009 Mar 19
0
DTMF tones mid conversation
Just to add.... P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055 P[ 1] --> DTMF
2011 Apr 05
2
dahdi and linux-2.6.38
Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question
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
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
2007 Mar 12
2
TDM-400, Polycom SIP phones, and echo problems
Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can
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
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
2007 Nov 28
2
DTMF not recognized on ISDN with Siemens -not IP- phone
Good day all, we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244 with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the office through ISDN lines and then get a possibility to join meetme conference. It works well except when customers are using SIEMENS phones (not IP): DTMF is not recognized. Does someone have an idea on what could be the problem with
2006 May 11
1
mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection
Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->
2004 Nov 22
2
dtmf tones during conversation
I have a * box running our house and on one extension we are getting spurious DMTF tones during conversations. It only happens on one of the 3 FXS ports and it's the one w/ a cordless phone on it. At first I thought someone was being careless and just hitting a button on the other end of the line, but it's happening too much for that... Has anyone run into this before? -- -M There