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

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.

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/.

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)

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

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

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

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:

Hi, I have connected a TDM400P to my asterisk, I have enabled DTMF CID, the data is arriving to the asterisk but asterisk isn't interpretating it: its my full log: 1. Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0 2. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Starting simple switch on 'Zap/4-1' 3. [Mar 10 16:26:03] VERBOSE[9274]

We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the '9' comes in and gets queued, but it doesn't start

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 have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]