similar to: AGI: Dial and Recording my own CDR

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, 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}

Here's what I'm doing: 1. I set __FOO=0 in the current context. 2. I use Dial() and the M(testvar) macro call. 3. The macro-testvar can see ${FOO} just fine. 4. the macro-testvar modifies FOO. I tried: Set(__FOO=1) Set(_FOO=1) Set(FOO=1) 5. When the Dial()ed call hangs up, and the original context continues, FOO is still 0.

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

I'm building an app that will do the following: 1. Force the caller to record their name. 2. Dial the party to call. 3. Play a short menu: 1 = Accept Call 2 = Decline Call, go to VM if available 3 = Accept Call forever, never ask again 4 = Decline Call forever, block number, get rid of caller 4. do things based on that choice. I'm

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

I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:

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 List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

I try to track down an error that causes that Astcc just reports the time, but not the costs. I could narrow the problem down into this sub routine: sub calccost() { my ($adjconn, $adjcost, $answeredtime, $increment) = @_; eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment }; my $cost; print STDERR "Adjusted time is $adjtime, cost is $adjcost with

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'm developping some AGI in perl (5.8.6) on i386 using Asterisk 1.0.5. I want to get some variables such as DIALSTATUS and ANSWEREDTIME after a $AGI->exec("Dial", "dial_string"); but here is what i get actually: DIALSTATUS= DIALEDTIME=ANSWER ANSWEREDTIME=18 I searched the archives and saw that $AGI->verbose could mess the access to variables, but I don't use

Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME. I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10 because, now, if ANSEREDTIME =~ 15.9, it become 15! it isn't correct How can I have a rounded ANSWEREDTIME ? Where have I to manipulate the sources? thank you -- Francois -------------- next part -------------- A non-text attachment was scrubbed...

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

Hi All, Asterisk 1.4.12 on CentOS 5 I'm trying to increment an AstDB key with the length of the last outgoing call. Here's what I've got for "01" UK geographical numbers: exten => _01.,1,Dial(${UKGeographical}/${EXTEN},,g) exten => _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME}) exten => _01.,n,Set(CALLTIME=${DIALEDTIME}) exten =>

Here i am doing a dial command from a php agi... EXEC DIAL H323/123456789@xx.xx.xx.xx:1720|40|HL(585000:61000:30000) But asterisk is not disconnecting the connection after 585 secs... the result is ... answered time is 1926n but thing is time out is working some time and some time not.... LOG: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: callcard.php: string(111) "app_callingcard:

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

I'm working on a project that involves Queues with Agents that are at home with a PSTN phone number, NOT connected via SIP phones. In the queues.conf it clearly states that only the SIP driver supports "In Use" detection of making members of a Queue available or unavailable. I've not yet figured out the best way to handle this. Currently I've got a macro that is executed

No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58", "") in new stack -- Executing