similar to: problem when exiting from "record file" function without pressing the escape digit

Hi All, I know that this may not strictly be an asterisk issue, but if anyone has any ideas on this prob it would be appreciated. I have a working asterisk setup using Suse 9.1, a BT ISDN BRI card (Fritz?), sipura 3000 and a few cisco 7940's. I wanted to do add faxing into the equation, so I thought I would try and setup the following: PSTN->asterisk->hylafax->pc At the moment

Hi, I think this is one area that needs to be developed. I am curently implementing a system for my home so cannot really justify the cost of financially supporting the development of this when all I really need to do is buy a telephone extension lead for my existing fax modem!!! I am more than willing to devote some testing/documentation time (I am not really a programmer) if that helps.

Dear WizaRds, unfortunately, I have been unable to replace the '99' and '999' entries in library(UsingR) attach(babies) as definitions for missing values NA, because sometimes the 99 entry is indeed a correct value. Usually, or so I thought, NAs can easily replace a, say, 999 entry via mymat[mymat==999] <- "yodl" in a matrix or data frame. Alas, the babies'

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

Hello folks. First of all, sorry for my English :-) I want to build dahdi-2.0.0 rpm from source (i have to use this version, because OpenVox A1200p driver works only with it). I've made some changes in .spec file (added one patch and one source section) and trying to build rpm: rpmbuild --define with-udev -bb SPECS/dahdi-linux.spec Executing(%prep): /bin/sh -e /var/tmp/rpm-tmp.86801 +

I have a master record called ''project'' and a child record called ''agycode''. Obviously, agycode has a project_id FK. I wish to make the "descr" field unique ONLY within the ''project_id'' ''scope''. Here are the key pieces of information Agycode fields (id, project_id, descr) Here''s the declaration in the

Dear List; For each call (in specific case), I need to do a record and save in a spearated file, so I am thinking the best thing is to save based on the time. Monitor(wav,Record1,m) So, how can I make the file name to be based on the current time (which is changed always, or based on the some unique paramter (related to the call it self). Any advise? Regards Bilal

I am having a problem with "record_file" working properly depending on when it is called -- basically if it is called immediately upon a call, it acts like it does not hear anything from the callers phone (yes, my phone is setup properly and functions fine otherwise)... if I do a "background" or "festival" command before calling it, it works fine. Details below:

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps

Hello, I''m wondering if someone might be able to offer me a solution to the following problem that has so far stumped me. I''m trying to get the distinct months (and years) from a date field to display as a list in a view. For example, there might be a number of records stored with dates in the table ''headlines'': name date record1 21-09-2008 record2

Hello, I'm using Icecast over 10 years, so let me first say *Thank you to all developers* of Icecast! I have a problem using a mount with relay and a fallback. The option <fallback-override> is not working. I tested two different configurations: _Configuration A:_     <mount>         <mount-name>/Stream-096k.mp3</mount-name>        

Dear All, I seem to have stumbled across an AGI problem; I have written an AGI Script (bottom of this email); The script does the following; Makes a CDR entry when called Records the call Updates the CDR Finds a corresponding DNIS from the SMDR table (captured via a serial port logger) Matches up the record and updates the CDR. The script works perfectly in my test lab and has been doing so

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.

Hello, I want a caller who is waiting in the queue to be able to exit this queue (and the waiting) by pressing a digit. I read in the wiki : /Context// //; A context may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context.// //context=<context>// //This is the

Hello, I'm using Icecast over 10 years, so let me first say *Thank you to all developers* of Icecast! I have a problem using a mount with relay and a fallback. The option <fallback-override> is not working. I tested two different configurations: _Configuration A:_ ??? <mount> ??????? <mount-name>/Stream-096k.mp3</mount-name> ???????

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

Hello list, Three questions on dialplan commands/functions: - cmd BACKGROUND: a) If i just want to allow user to send DTMF ONLY while the message is played (with no additional time after ), does TIMEOUT RESPONSE = 0 make sense or this action produce some border effect ? b) if YES (border effect = stupid me), may i use an AGI with 'record_file' and 'get_data' to do something

Hi, I've got a setup whereby calls come into the asterisk server (1.2.7.1) over a IAX2 trunk and into a dialplan that launches a php AGI script: [live-full] exten => _X.,1,Set(TIMEOUT(absolute)=0) exten => _X.,2,NoOp(${EXTEN}) exten => _X.,3,DEADAGI(live-full.php) exten => _X.,4,Wait,2 exten => _X.,5,Hangup The script is using phpagi-2 from http://phpagi.sourceforge.net/ and

Hi guys I am facing a problem in recording voice on Hangup. I am using php agi class for this purpose. Currently its voice message is being recording when i used to press 1. For this purpose i am using record_file() function with its respective parameters. Is there any way that i can be able to record my voice message on hangup (don't want to send 1 as DTMF to asterisk.). Can you guys guide me