similar to: Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

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:

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on.... This looked hopeful for a moment until I realised that it doesn't do DTMF:

On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. > That's probably not something you can hack away at with simple > Asterisk

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

using a zap fxo and zap fxs card how can I set up caller announce? like this. 1 call comes in and a prompt asks the called to identify themselves. 2 the system would then put the caller on hold and pick up the FXS and play the message for the users prompting them to hit 1 to accept the call and have it connected or hit 2 to dump the live caller to voicemail. Can this be done with * Dave

== Parsing '/etc/asterisk/adsi.conf': Found -- Accepting call from '890003' to '185' on channel 27, span 1 -- Executing Answer("Zap/27-1", "") in new stack -- Executing Record("Zap/27-1", "soundexampless:mp3") in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128

Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [*40 at liberado15:15] Record("SIP/1201-083453c8", "/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3") in new stack ??? -- <SIP/1201-083453c8> Playing 'beep'

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

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Hello, I am trying to install my login and search forms as default parts of the layout. This is the code I am using: <% form_for :user, :url => {:action => ''authenticate''} do |f| %> <p>Username:<br /><%= f.text_field :username, :size => 30 %></p> <p>Password:<br /><%= f.password_field :password, :size => 30

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

I don't see a method to detect the success or failure for the Record CMD. I'd like to know the reason why the recording ended Am I wrong? exten => recordmsg,1,Noop() exten => recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180) Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello puppet gurus, maybe one of you can help with this. We have a bunch of servers that are managed by puppet, but would like to make a single, small change to a text file only on newly built machines. This text file is controlled by a module that is referenced in several places in our current configuration files. So the question would be, is there a straightforward way to have this new config

Some unsuccessfull attempts to make console calls working. If a sip phone is called, the other side will hear nothing. If I try to record some sound the application will not finish. There is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)" is used in the dial plan. After hangup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format:

Hi All... I'm trying to use the record() app and it complains that it can't open it's file because permission was denied. I'm running the released Asterisk on Debian Linux. The target directory is workd writable. Here is the relevant part of the dialplan: exten => 1,1,Playback(leave-message) exten => 1,2, Record(/var/local/whois-messages/whois-${contactid}:wav|6|120)

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

Hello, I've been working on an * dialer application, whereby a requirement is that if no one answers the call, a message must be left on voicemail. I've been using the record(tmp.gsm) function with silence detection enabled to wait for the greeting to finish before speaking. However, on voicemail systems where you can interrupt the greeting with a pound (#) key to access your voicemail

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: