similar to: Shorten time between DTMF

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

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?

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

Suppose a user submits a url: http://www.nyt.com/education/2345545. How can this be shortened to a cleaner url, like nyt.com? -- Posted via http://www.ruby-forum.com/.

Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone -> chan_dahdi g1 -> asterisk -> can_sip -> SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2.

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

On Wed, 7 Feb 2001, Josh Coalson wrote: > Mark, if it's possible, can you do me a favor... > Try encoding the album as individual tracks and > compare sizes. The reason I ask is because of > the way FLAC frames are numbered in the frame > headers (if you check the format page you'll see > what I mean). Sorry, I don't :( I've split into individual files and

had to fix the subject... was getting under my skin! yeah, could you put up the FLAC version of the worst track that is less than 20 megs compressed? (I'll have to grab it with a 56k modem). by worst I mean the one where shorten beats flac by the most. also: 1. what version of shorten are you using? 2. what command-line options for flac and shorten did you use on this track? thanks, Josh

well, I took a look at the files. from my knowledge of shorten there are two things it does that flac doesn't do: 1. it estimates the mean of the signal for each block, subtracts it out and stores it separately. but this is pretty useless for the predictors that shorten uses as they are pretty insensitive to the mean (try different values of -m from 0 to whatever and note practically no

Hi List, I am using Asterisk 1.6.2.18. One strange problem come into my knowledge after using this version of asterisk. Without pressing any digits or key from my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and

When offset + length == exts->end the code which is shortening the extents is subtracting 0 from length. And it is not optimized out. Signed-off-by: Martin Kletzander <mkletzan@redhat.com> --- server/extents.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/server/extents.c b/server/extents.c index d3d1a15ab97c..c422491601f0 100644 --- a/server/extents.c +++

I've found a wav of an entire album that shorten compresses better than flac. Is this supposed to be possible? wav 388,374,044 flac-6 239,109,203 shn 236,457,697 Cheers. Mark Powell - UNIX System Administrator - The University of Salford Academic Information Services, Clifford Whitworth Building, Salford University, Manchester, M5 4WT, UK. Tel: +44 161 295 5936 Fax: +44 161 295 5888

Hello, I'm using Linux for about two months now. Even before switching to Linux I have heard of Wine. I got my system running relativelly stable (I use Slackware 10.2, kernel-2.6.14 and KDE 3.4), but still have some issues - sound being one of them. But that's my problem. I'd like to contribute to the wine project somehow. Perhaps starting as a test user for some application.

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time

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: