similar to: Random DTMF tones generated from speech

Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB

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

Hi, Is it possible to "regret" blind transfer while its ringing (not answered)? Thank you! Best regards HB

Hi! I tested two different implementations of the robust MCD estimator: cov.mcd from the MASS package and covMcd from the rrcov package. Tests were done on the hbk dataset included in the rrcov package. Unfortunately I get quite differing results -- so the question is whether this differences are justified or an error on my side or a bug? Here is, what I did: > require(MASS) >

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

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

Hi Any solution for connecting Asterisk to Skype without using fsx/fxo hardware ? Best Regards HB Norway

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

Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? -- Your life is like a penny. You're

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

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

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

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 in the myshare share. Here are the definitions. [myshare] comment = root directory path = / #fake oplocks = yes writable = yes printable =

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

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

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

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

Hi. I can''t find any documentation on the specific selectors for tc-filters -- what documentation I have says they are in Polish in a file called selectors.html -- is there anything around in English to see those? Thanks. -- Your life is like a penny. You''re going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com

Hi. I want to use the new asterisk 1.4 with dahdi, but I would like to know the svn branches for the dahdi, so I can use them that way -- much easier to keep up with bug fixes, etc. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com

Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known