similar to: Asterisk 1.4.15, Solaris and record command

I submiited to the list last night, but it never showed up. Here we go again. I've tried building Asterisk 1.4.15 on Solaris based on instuctions here, http://forums.digium.com/viewtopic.php?t=5888. However, this is the message I get. This is Solaris on X86. Any ideas? [CC] stdtime/localtime.c -> stdtime/localtime.o stdtime/localtime.c: In function `localsub':

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

Hello, We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently we're using the AGI command "record file" which does allow us to specify which DTMF keys can end the recording. However we also need to know *which* key actually ended the recording. Note that only allowing # or * to end the recording won't work for us. Does anyone

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using

Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to

Just a quick and dirty thought, try the MONITOR application. Pseudo-code: Anchor-point PLAYBACK ("press or say") MONITOR (use the split audio files mode, not the mixed - this way you can roughly separate which side did the "talking") READ (audio file "1 to 5", try to grab one digit) STOPMONITOR IF (READ variable timed-out, send the incoming half of the monitor file

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there

Hello list, I wonder if this is possible with Asterisk: - While talking through Asterisk, I would like a client to start recording a call by typing, say, #99# I know it is possible to do it using an external monitoring application, but I want to know if it's possible to have Asterisk silently monitoring an on-going call and responding to DTMF tones within it. How do things like call

On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote: > Oh, what a good idea! That's exactly the kind of lateral thinking I > was hoping someone would come up with. > > I thought it was called MixMonitor, and tried to wrap my head around > it but couldn't. MixMonitor is related, but different (and as the name suggests, automatically mixes the two channels, so I

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2:

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

Hi all, I am using an asterisk-1.4.13 connected to our carrier via SIP trunk. I use rfc2833 as dtmf detection method. After upgrading to asterisk-1.4.15 our system would not detect dtmf from a caller from PSTN anymore. When investigating the SIP traffic at call initiation I realized that in the SDP message asterisk is no longer offering the telephone-event/8000 capability. So the carrier does

I'm attempting to set up call recording with Asterisk. Using automon => *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or

I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

Hi... In my dialplan, I have the following: exten => s,1,Background(${RECORDING}|m) exten => s,n,Voicemail(${DID_NO}) exten => 0,1,Voicemail(${DID_NO}) exten => a,1,VoiceMailMain(${DID_NO}) exten => h,1,Hangup In version 1.2, when I hit "0" during the playback, I will be directed to voicemail. But in verison 1.4, the call hangs up. [Jan 24 16:05:37] DTMF[5754]:

Hello, Is it possible, that during the call one side , for examples clicks the button on the web, and this call starts recording? It's possible with asterisk feature automon and DTMF. So it is possible to start recording the channel using AMI or ... ? Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"