search for: record_fil

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" comma...

...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 => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/confbridge/${TL_PHONE_CALL_ID}.wav)) same => n,ConfBridge(${TL_PHONE_CALL_ID},default_bridge,TTM_caller,TTM_caller_menu) Here is the log from asterisk 13 -- Executing [s at TTMConferenceTalker:6] ExecIf("SIP/sbc1-00002426", "1?SET(CONFBRIDGE(brid...

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 me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 "Shoot for the moon. Even if you miss, you'll land...

...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 for months. I have moved the script over to the production environment and the script stops after; $AGI->record_file($fname,'gsm',9,-1);... Since the test environment was different to the production environment; I upgraded my test environment from Asterisk V1 and RedHat 9 to V1.2.4 and CentOS. Now in the test environment my script stops after; $dbh ->do("Update CDR SET Recording='0',GS...

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 everybody, I've been working on a project which records the voice of the incoming call. I use record_file function of asterisk as described below: RECORD FILE <filename> <format> <escape digits> <timeout> [offset samples] [BEEP] [s=<silence>] filename: record1 format: wav escape digits: # timeout: -1 offset samples: 0 BEEP: 1 silence: 3000 *Please read the scenario for...

...call my conference, I see this error in my logs: ERROR: Function DENOISE not registered here is snippet from extensions.conf ... same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes) same => n,Set(CONFBRIDGE(bridge,record_conference)=yes) same => n,Set(CONFBRIDGE(bridge,record_file)=/home/asterisk/file.wav) same => n,ConfBridge(1000) same => n,Hangup() and I have "denoise = yes" in confbridge.conf: how do I fix this? What do I have to do to "register" denoise ? thanks,

...mands/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 like that ? ( timeout is in milliseconds in this case => if '0' is a dangerous value, perhaps i could use timeout = someveryverylittlevalue 100/200 ms The goal is to have a 'juke-box' playing messages (not only sequential order:...

...e is related to asterisk rather than the script itself. As the script is launched using DEADAGI I expect it to carry on after the channel has been hungup (to save the results of the user input in this case) which works unless the users are leaving a voice message at the time. The script uses "record_file" and records ok if the user ends the call with a keypress (#) but if the user hangs up once they have finished leaving their message the script exits immediately rather than carrying on: Nov 2 11:45:57 VERBOSE[24262] logger.c: AGI Rx << RECORD FILE /ivr/recordedtemp/1162467957 wav &qu...

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 towards its solution. Thanx in Advance -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Ne...

...uld like to know if something I am trying to do is possible. I currently using 1.8 and would like to make recordings in wideband (16khz) I have an aastra 6739i which supports the g722 codec. I made an agi application in which I would like an user to record a promptlist. I use the agi command record_file. Which format setting do I need to be able to record in wideband? Tried: wav, gsm, pcm. Nothing seems to give me the result I desire. Hope someone can help me. Kind regards, Henry Dogger Telecats BV -------------- next part -------------- An HTML attachment was scrubbed... URL:...

Dear friends, Need to know filenames of conference recordings in Asterisk 11. Besides directory scanning the recordings could use CEL: Filter MySQL rows with eventtype equal CHAN_START and channame like ConfBridgeRecorder and then get the eventtime field and convert to timestamp to complete filename(s). Would you suggest any other approaches? Thanks. -------------- next part -------------- An

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it