similar to: Asterisk 13.4.0 - mixmonitor only records one side's perspective

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

Hi, I am using this dialplan to record incoming calls: ..... exten => 3331122,n,Set(MONITOR_FILE=${RECDIR}/${UNIQUEID}) exten => 3331122,n,MixMonitor(${MONITOR_FILE}.wav,b) exten => 3331122,n,GoSub(stdexten(${Ext1007})) exten => 3331122,n,Voicemail(1007 at default,) exten => 3331122,n,Hangup() The problem is it records all incoming calls include those with the disposition of

Hi, Below link is the script which i found while surfing, this script basically converts your voice file to flac format, where the file is reduced to 50%. http://legroom.net/files/software/convtoflac.sh The quality is really good, I tested. this... In large production environment this script can be used, only challenging part, please make sure the CPU usage is within the limit while

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Alrighty Gents, let's see if any of you have encountered this one...Variables losing their value...I'm setting a variable with four underscores (used to be two, had same issue) so it can be inherited by child channels, and then the next line in the dialplan I use it but it appears to be empty...I've googled and found nothing stating this kind of weirdness.. Asterisk 1.8.2.2 (upgrading

hi all , please need help for an asterisk version 1.4.21.2 i created a write func odbc list records files in sql table: [R] dsn=connector write=INSERT INTO ast_records (filename,caller,callee,dtime) VALUES ('${ARG1}','${ARG2}','${ARG3}','${ARG4}') prefix=M and set it in dialplan : exten => _0X.,n,Set( M_R(${MIXMONITOR_FILENAME}\,${CUSER}\,${EXTEN}\,${DTIME})= )

I am having a problem setting the caller ID that shows when I make an outbound call over my PRI line. If I make a call from a SIP phone registered with the Asterisk box the PRI is connected to the correct ID shows on my cell phone. If I make a call from an IAX trunk connected asterisk box calling the same number as call one and setting the caller ID to the same number as call one the caller ID

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hi all I need to pass a parameter in a thread-safe manner to the Queue pickup macro. This is to know when (and who) picked up an incoming call to a queue and log that to my back-office system with a CURL to a HTTP endpoint. However, the Queue application does not appear to allow passing of parameters to the called queue pickup macro. E. g. non-working code is: [queuetest] timeout = 60 retry =

Hi Marie Thanks! I was just worried about thread safety if I had to use a global variable, e. g. it might be set to a value by one call (since I'm using the same global for every incoming call to transfer the accountcode gotten from my HTTP endpoint to the same macro, and there can be several calls simultaneously all inserting HTTP-sourced values at more or less the same instant) and then

On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: > Hello Carlos, > > >> We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

On Thu, Jun 26, 2014 at 3:39 PM, Robinson, Paul <Paul_Robinson at playstation.sony.com> wrote: > The main motivation for turning it off is that no known consumer (debugger) > took advantage of it. > > Turning it on does more than slightly increase the object file size, it can > cause the same source line to be listed multiple times in the .debug_line > table (with different

Hello, I'm compiling Asterisk from source on Debian systems. I'm currently writing a script I'm planning to launch when upgrading from one Asterisk version to another one within the same class (from 13.4.0 to 13.12.0 or from 13.12.0 to 13.8.0, for instance). Reading [1], I thought the following would work: cd /usr/src/asterisk-13.4.0 ./configure make make install ... cd

Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone volume. I can see that the