similar to: wrong time retrieved from system command

I am using the latest SVN version 1.2 of Asterisk When I attempt to test the output of certain variables, for use in file naming etc, certain key ones appear to fail. exten => 5555,1,NoOp(${EPOCH}) Returns -- Executing NoOp("SIP/200-638c", "1141352935") in new stack exten => 5556,1,NoOp(${TIMESTAMP}) Returns -- Executing NoOp("SIP/200-8cc9",

http://bugzilla.mindrot.org/show_bug.cgi?id=119 ------- Additional Comments From stevesk at pobox.com 2002-03-31 05:22 ------- what is the timeout issue? timeouts can be caused by filters that maintain connection state and have timeouts; or TCP keepalives or ??? this should probably be "invalid" ------- You are receiving this mail because: ------- You are the assignee for the

I'm trying to declare a few date-related global variables to ease my dialplan. When I declare the following in the [globals] context of extensions.conf, I get unexpected results: YEAR = ${STRFTIME(${EPOCH},,%Y)} MONTH = ${STRFTIME(${EPOCH},,%m)} DAY = ${STRFTIME(${EPOCH},,%d)} TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} If I evaluate these variables in the dialplan later, using exten

Hi all Is there anyway i could get in the dialplan the amount of time a caller waited in the queue before exiting? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160518/b3b082aa/attachment.html>

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 bridged so why would Asterisk create a file? Is there a way to avoid getting those empty

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

On 1.6.1, I must be losing my eyesight: [internal] include => outbound-pstn ............. include => meetme ; 2663 include => setup-meetme-conf-room ; 6000xxxYYYY [setup-meetme-conf-room] exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" ) ........ CLI: -- Starting simple switch on 'DAHDI/1-1' [2009-05-17 14:54:49]

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

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"

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards!

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

Hello, I need to use the ${DATETIME} macro inside the filename saved by Record, but the colons (':') used in the time interfere with the command (everything after the colon is interpreted as the format I wish to save to): My command is: Record(/path/to/voicemail/${EXTEN}-${DATETIME}-${UNIQUEID}:wav) I need some function to escape the colons inside DATETIME... I'm sure it's

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I need help need recording all call for my pbx but i am a novato in asterisk my confi for record is: exten=>_NXXXXXXXXX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}) exten => _NXXXXXXXXX,n,MixMonitor(${CALLFILENAME}.gsm,m) exten => _NXXXXXXXXX,n,Dial(${TRUNK_CLIENTE}/${EXTEN}) -- Bayardo S?nchez Garc?a Web Developer - Internet

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

Hi all, What's the best method to send automon call recordings (*1) to the voicemail box of the Asterisk user? Do you have to trap hangups, etc, or is there some global variable that can be set? Thanks! S.

I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far.

You can't set the variable in globals? I don't know if functions work in globals, but it is worth a try. [globals] LSESSION=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} On 8/16/23 20:39, Federico wrote: > I used to use the local channel to create a global variable > > (dialplan) > > [default] > > exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d

I can't get a test T.38 fax between 2 1.6.2 machines, using app _fax and spandsp pre17 and 20100501. The machines can't seem to get connected. send side extensions.conf: [fax-tx-test] exten=>s,1,NoOp(Context fax-tx-test) exten=>s,n,SendFAX(${FaxFile}.tif) exten=>s,n,HangUp() exten=>h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} FAXMODE: ${FAXMODE})