similar to: Inter-Digit delay when dialing out

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 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

Hello, I am trying to bootstrap a function that extracts the log-likelihood value and the nls coefficients from an nls object. I want to sample my dataset (pdd) with replacement and for each sampled dataset, I want to run nls and output the nls coefficients and the log-likelihood value. Code: x<-c(1,2,3,4,5,6,7,8,9,10) y<-c(10,11,12,15,19,23,26,28,28,30) pdd<-data.frame(x,y)

Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers

I'll rearrange a little and respond: --- Mark Powell <M.S.Powell@salford.ac.uk> wrote: > Also, when flac takes input from stdin it fails to > fill in the wav size > fields correctly, whereas shorten has no problems > with this: i.e. >... > You can see it puts a data chunk size of zero in > there. > OK, this has been fixed in CVS. > Flac refuses

Hi List, I've had some issues with some VoIP providers where either: 1 - There is massive PDD but finally the call goes through 2 - There is massive PDD but the call gets rejected anyways I was wondering if there was a way to automatically detect such behaviors when it happens (maybe with a script or something) so I can take the faulty providers out of the routing and maybe automatically

I see there is the DigitTimeout application that sets the maximum time between digits before asterisk will interpet. Is there any way to control the minimum? We are having problems with incoming calls on our X100P where callers try to dial 10, but the 1 gets detected twice and they end up on extension 11. Thanks Mark Farver

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

I have an installation that has analog phones connected via T1 channel banks. I'm getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns:

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __________Asterisk fxo ---- line -----| -----------------Analog phone The analog phone rings immediately when calling, while asterisk shows the message

Hi, I am using Asterisk 1.4.17 for my C4 routing but I am experiencing a high pdd of around 20 seconds. Could you please help me to reduce it and what could be the reason. Thanks Abid Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110509/163192a7/attachment.htm>

Anyone got a way to pull the Source and Destination IP addresses for a call so they can be logged in a CDR and or CEL? Also anyone got a slick way for registring PDD on calls? Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120213/7ef4ad08/attachment.htm>

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 THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

Hi everyone I would like to enable unprivileged users to share only certain directories using SFTP without acquiring root, without setting capabilities using public-key-based forced commands. In another use case unprivileged users could write scripts that evaluate "$SSH_ORIGINAL_COMMAND" and then either execute sftp-server in a jail "$SSH_ORIGINAL_COMMAND" after

Hi, After using Asterisk with a SIP hardphone for a couple of weeks I've just installed a TDM400P card. My hardphone - a 7940 - allows me to use a dialplan to decide when a particular extension is complete and automatically trigger dialing. This works well with my internal extensions, which are all of the form "Z00". When trying to dial these extensions from a handset

Hi, I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet card (with ports in PTP mode). I noticed a long delay when making outbound calls, more precisely between (taken from Asterisk CLI) "Called 1/XXXXXXXXX/s" and "mISDN/1-u43 is proceeding passing it to SIP/8-5486" I searched on misdn.org but found nothing. I'd like to understand if this delay is

I have a delay of "0" on SpeecBackGround, but when I enter DTMF, there's an almost-exactly five second delay before it returns. Where is this delay controlled? How can I shorten it? Is there a way to set the maximum number of digits to look for?

Greetings, Thanks for your time in reviewing the following role located in the RTP, NC area. Our client is a drug development platform technology company whose patent-pending intellectual property is poised to dramatically impact both drug discovery and development processes by accurately measuring the spectrum of biochemical changes and mapping these changes to metabolic pathways. The