similar to: Help with FUNC_MATH

John, >From looking at the wiki won't STRFIME just give me what I need based on the unix time that I put in? What I am actually looking to do is convert over from 12 hour format to 24 (unless strftime does just that and I don't kow what am I am doing?). On Thu, Feb 13, 2020 at 12:03 PM John Kiniston <johnkiniston at gmail.com> wrote: > Try using the STRFIME function

HOUR_SELECTED is going to be 1-12 On Thu, Feb 13, 2020 at 2:05 PM Don Kelly <dk at donkelly.biz> wrote: > Is HOUR_SELECTED a floating-point number (e.g. 11.99999999999)? If so, you > need to account for that in your comparison. > > > > --Don > > > > > > *From:* asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] *On > Behalf Of

John, That is correct. I am trying to figure out why Asterisk is executing the set part of the execif, if it's coming back as false. On Thu, Feb 13, 2020 at 2:10 PM John Kiniston <johnkiniston at gmail.com> wrote: > My Apologies Dovid, I think I misunderstood your request. > > You don't have the time you need to convert in the format of date string, > Instead you

Is HOUR_SELECTED a floating-point number (e.g. 11.99999999999)? If so, you need to account for that in your comparison. --Don From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dovid Bender Sent: Thursday, February 13, 2020 4:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help with FUNC_MATH Hi, I

Try using the STRFIME function instead of doing this by hand. https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME *%H* The hour as a decimal number using a 24-hour clock (range 00 to 23). *%I* The hour as a decimal number using a 12-hour clock (range 01 to 12). On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender <dovid at telecurve.com> wrote: > Hi, > > I have some dialplan

My Apologies Dovid, I think I misunderstood your request. You don't have the time you need to convert in the format of date string, Instead you have your users entering via DTMF when they want something to happen? On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender <dovid at telecurve.com> wrote: > John, > > From looking at the wiki won't STRFIME just give me what I need based

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried: ; strip leading plus sign same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} ) same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = ${CALLERID(num):1})

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten => s,1,NoOP same => n,Answer same => n,Set(filename=$PAGE) same => n,Wait(1) same =>

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

On Tue, 30 Aug 2016 17:56:35 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > Something like > > exten => 5555551111,1,Verbose(Door buzzer calling) > same => n,Set(toRing=) > same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN > USE"]?Set(toRing=${toRing}&SIP/user1) Failed. I checked the online docs and the syntax seems to

Imagine that you have this code: exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})})) If ${QueueName} happens to be unset, this will cause a warning: [Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187 queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an argument: queuename The obvious solution: exten => _X!,n,ExecIf($["${QueueName}" !=

Hello, we want to use something like same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) Problem is that result gives C=1) & Set(D=2) & ... Is there a possibility to use multiple AND or OR in such a way ? -- Daniel

On Thu, 1 Sep 2016 11:02:57 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > > [Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application > > 'ExecIf' for extension (unauthenticated, 5555551111, 3) > > > > Is there a module that I need to load? > > > > In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0. > >

Hello Some of our customers call with CID blocked. I'd like to save those numbers into a SQLite database using a command-line PHP script, so that I can... 1. Edit the CID name through a PHP web script which will just list all the customers in the database who have a phone number but no CID name set 2. Look up those customers' e-mail address from this database, and send them an e-mail

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >