similar to: Delay on DTMF with SpeechBackground and Vestec

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

I'm having a lot of problem with it recognizing "oh" for zero. I've tried both "o" and "oh". In one case, I just tried: $digit = o { out = "0"; } | fundamental {out = "2"; }; So I gave it a choice that was VERY far away from what I said. But when I said "o o o o o", more than 75% of the time, it had a bunch of

Okay, I see that there is no way to shorten the inter-digit delay when we dial out DTMF. The reason I need this is because when dialing a two-stage call, the PDD can be shortened dramatically while still working fine, and that opens whole new world of business. But many people confused dialing-out with receiving DTMF, which is not what we are concerned with. If it is impossible to configure the

Does it exist? Sending email to their support address appears to be a black hole. They reference a forum, but Google can't find it. I keep having problems in any grammar than has a an "o" for "zero": it breaks recognition anywhere NEAR it. For example, if I say "two o five", it gets recognized as "one o five".

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?

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

Has anyone noticed a delay in sending DTMF to zaptel devices? I have a T100P connecting to an NACT telecom switch. All calls are sent just fine, but there are 4 seconds of delay between when the channel goes off-hook & the digits are collected by the NACT switch. I also have some cisco 5300 boxes connected, but there is no delay. Any ideas? -g

I have a TDM22B (TDM400 PCI + 2xFXO + 2xFXS). One FXO is connected to the PSTN. When Asterisk places a call, it dials using DTMF. If I listen in on the line during the dialing, there is a roughly one second pause between the penultimate and the final digit -- proportionally much longer than the pause between the preceeding digits. This accounts for about 25% of the time taken to dial. Anyone

Hi, I'm am getting doubled DTMF on some digits with one of my providers who also uses asterisk. We're using SIP, with dtmfmode set to rfc2833, and the codec G.711. Once out of every five or ten calls, there are no problems, but more often than not, the DTMF is getting doubled-up on at least one of the digits of the extension dialed. I've tested with a CVS-HEAD from Febuary, and just

Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio?basically, like a podcast?and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

Hi, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I've had a look on my Account page on the Digium website but it only shows the Language Pack, and I can't do anything with this either. Can anyone point me in the right direction please? Thanks Lee

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 All, I am trying to call my own service through Asterisk and the DTMF is not recognized . I also noticed the following issue, the phone rings for about 8-9 times before the line is picked up but when it is picked up it seems that our system has picked up the call much earlier, I could just not hear anything except the ring. that means other system picked UP a call and my SIP phone still

Hi, I am using zap channels, and by using sendDTFM application, I can control the duration between two DTMF digits, but I can't find a way to control the duration of the digits themself. Did search on the Internet and found out that I can change it in the asterisk source files and recompile asterisk. Wiki also says that it can be controlled using toneduration option in zapata.conf, but it

Hi listers, I'm a 1.4 holdout who uses mostly Suse platforms. I bit the bullet and installed a Centos box to get Lumenvox up and running, but now see this "new" offering of Vestec that supports OpenSuse and Windows in addition to the Platforms supported by Lumenvox. Anybody out there working with Vestec that can give me a heads up on how it works or doesn't?

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in

Hi List, We have a major issue while broadcasting DTMF using meetme application. We are sending DTMF to asterisk using SIP INFO message with duration 160. INFO sip:xxx at xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060 From: <sip:xxx at xxx>;tag=43 To: <sip:xxx at xxx>;tag=9753.0207 Call-ID: xxx at xxx CSeq: 25634 INFO Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2\r\n

Hi, I'm having trouble to figure out how I could implement this feature : "When on call with a contact, local operator would dial a sequence which would park the remote party to a specific parking slot, among the hundred of existing slots. (to each extension, a single specific parking slot is attached and there are too many extensions to dedicate BLF or short DTMF sequence to each)