similar to: DIAL() application 'g' option

Hi All, Asterisk 1.4.12 CentOS 5 My ISP account includes nearly 500 minutes of VOIP calls per month but the service is expensive for unbundled minutes. So I'm trying to find a way to keep an accumulated total of calls made through that trunk so that I can automatically switch to a lower-cost provider when my bundled minutes are used. The plan is to store the accumulated time in AstDB and

Hello I've tried a bunch of things, but still get errors/warnings when using the IF() function: ============== TEST #1 exten => h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)} ]?${CALLTIME}.wav)}) [Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax IF(<expr>?[<true>][:<false>]) ============== TEST #2 exten =>

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 =>

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

Hi R folk I have a distance time graph for a locomotive and at various times different events occur on board the loco. I want to put a vertical line on the speed time graph for each event, but I want to colour each different kind of event differently to see visually whether there's any pattern to these events happening. I could just create a vector of colours and use abline which is easy

Hi folks -- I have a FastAGI Perl script running, handling calls. It works great. At one point I have a Dial() command. If the called party hangs up, Dial() returns 0, and when I call my own recordCdr() function using the channel variables ANSWEREDTIME, DIALEDTIME and DIALSTATUS, everything is fine. However, if the called party picks up, and then the dialing party hangs up Dial() returns -1,

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}

Does: rainbow(3)[1] rainbow(3)[2] rainbow(3)[3] ... solve your issue? B. > On Jun 8, 2017, at 8:20 AM, WRAY NICHOLAS <nicholas.wray at ntlworld.com> wrote: > > Hi R folk I have a distance time graph for a locomotive and at various times > different events occur on board the loco. I want to put a vertical line on the > speed time graph for each event, but I

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

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

Hi All, I'm stuck and need some help. I have installed the Asterisk CDR Analyser Version 2.0.1. It mostly works except for the CDR Report. I get the following error even though it lists the CDR details. Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day, sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE UNIX_TIMESTAMP(calldate) >=

Hi, I'm currently using a cheap SIP provider for outbound calls. I do have 6 channels to them. In their terms of service there is the following limit: The total duration of calls during one single day should not exceed 24 hours or we do have the right to terminate the contract...blah blah What is the best way to use this provider as long as we are below let's say 22h in a single day

Is anybody using http://freevoip.gedameurope.com ? I've a problem with dial out. -----from help menu----------- In the help setup menu there is an example : exten => _683XXXXX,1,Answer() exten => _683XXXXX,2,Dial(IAX2/6840369/${EXTEN}) The second allows you to call FreeVoIP members by dialing 683XXXX where X is a number from 0 to 9. ------end from help menu--------- I think the first

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

Hi All, Asterisk 1.4.12 on CentOS 5 I'd like to be able to look up each incoming CLI to retrieve an associated name, if available, and then pass that to the extensions so that they can see both the name and number of the caller. I'm not after LDAP or anything else maintained externally, just a contact lookup for my system. I suspect that Astdb could be used for this, as could a

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :-

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? Please help me, where can I add SipAddHeader() in

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? regards, Asif

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph