similar to: problem with dtmf detection in asterisk 11

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

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

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

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

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, 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

On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael <venefax at gmail.com> wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call

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

Rers: I have some ecological data (stream velocity vs. % cover of submerged weeds) that shows strong evidence of a thresholding step-function, e.g. below some velocity, % cover ranges from 0% to 100% (with no apparent relationship to velocity within this range of velocities), but above a certain "threshold" velocity, the % cover does not appear to exceed, say, 10%. There are good

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

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

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

Hello. I've just started using R and am trying to figure out if the two codes snippets below have the same output gBest<-floor(runif(popsize,min=1,max=top)) velocity[i,j]<<-.4* velocity[i,j] + 1 * runif(1) * (pbestsVar[i,j] - popVar[i,j]) + 1 * runif(1) * (archiveVar[gBest,j] - popVar[i,j]) and for (i in 1:popsize) { for (j in 1:maxvar) { gBest<-sample(top,size=1)

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

Hello, does anyone knows this predict is not resulting? # lm predict dfTestes3sitesCriptic$Velocity_corrected <- ifelse (dfTestes3sitesCriptic$Season == "A" & dfTestes3sitesCriptic$BeforeAfter == "Before", (dfTestes3sitesCriptic$Velocity * mVel3ABefAfter), (ifelse (dfTestes3sitesCriptic$Season == "Sp" & dfTestes3sitesCriptic$BeforeAfter ==

I have constructed a Trellis style xyplot. lengthf <- factor(length) xyplot(SLI$velocity ~ SLI$width | SLI$lengthf, layout = c(2,7), xlab = "Width (cm)", ylab = "Velocity (m/s^2)", col = "black") This produces a lovely little plot. However, the grouping factor(lengthf) isn't in the right order. My values range from 2-28 and the 2 graph on the bottom left and

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