similar to: Running AGI script if condition met?

I would like to hire someone to automate my asterisk for hosted PBX service for fetures like user signup, adding money and call bridging Please contact me offline at ted at 1ezphone.com ----- Original Message ----- From: "Philipp Kempgen" To: "Asterisk Users" Subject: Re: [asterisk-users] Running AGI script if condition met? Date: Thu, 06 Dec 2007 05:11:24 +0100

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

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 Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

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 =>

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

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})

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

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

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Got a strange situation [ext-queues] ... exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X POST --header "Content-Type: application/json" --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "

Hi, I have some dialplan code that is trying to convert 12 hour time with AM/PM to 24 hour format. The code has something like this: Exten => 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)})) Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 they are selecting PM. If the time is from 1PM to 11PM then I want to add 12

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}" !=

We're having an issue where we call ReceiveFax in a context that includes a hangup extension and half the time dialplan execution doesn't continue after the fax is received successfully. Am I missing something simple here? Below is a sample call where this happened: The last log line for this channel/call is: [Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel

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

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

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for