similar to: Dial from AGI [MSG]

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Hello List, This is my first message herein. I was playing around with System() and AGI() and found out something I cound not determine my configuration error. I added before.agi and after.agi to the agi-bin dir. Tried to make before.agi get run before the dial call and after.agi be run after. Only the first priority (step 1) gets executed. Here follows some relevant part of the tests: On

Hi, I have come across an odd problem. Basically I am transferring a call to an agent. The agent is logged in and set as paused. In order to find which agent to call I am using a fastagi script to just set a variable. When it falls through the agi script and dials the agent (using the variable) it doesn't connect the call properly to the agent. I get the beep but no audio (along with

Dear All, I seem to have stumbled across an AGI problem; I have written an AGI Script (bottom of this email); The script does the following; Makes a CDR entry when called Records the call Updates the CDR Finds a corresponding DNIS from the SMDR table (captured via a serial port logger) Matches up the record and updates the CDR. The script works perfectly in my test lab and has been doing so

Am using perl AGI to invoke the dial command thus: $AGI->exec('Dial',"$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002)"); What I expected that this will do is: 1. call the number using the string $numtodial2 - works OK 2. Set call limit to $maxcall and play a message $msgtime milliseconds before the call - works OK 3. On connect of the call send it to the macro conn

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Hi. I am using #include to include a file in extensions.conf. I have an agi perl script which modifies the #included file and then forces an asterisk reload with 'system("asterisk -rx reload")'; After the reload I use set_context, set_extension and set_priority to tell asterisk where I want it to carry on from which is line 2 of the following. exten => *80,1,AGI(bla.pl,

Hello * users, I know that passing variable in the AGI script is by exten => _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being passed and simple_c_prgm is C code Now how will I receive these variables within C code ? Is it by the same way arguments are passed in command line to C by using argc and argv or there is more to be done than that? Thanks Regards -- Arpit Mehta

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

Hi I was wondering if its possible to make Dial command bridge two channels and after bridging bypass asterisk, so that the voice doesn't need to pass through my asterisk server. For e.g., I have a user dialed in and he verifies himself and then dials an international extension, after the call connects I don't want the call to pass through asterisk server anymore. Is there any command

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

I'm writing an AGI script and want it to dial a number on a channel connected to the PSTN. It would look something like this (pseudo-code follows): if ($a){ dial("8005551212"); }else{ dial("8665550000"); } The part I can't seem to get right is the "dial" function. I tried to mimic the dial plan like so sub dial($number){ print

I would like to run an agi script (to calculate the cost of a long distance or international call) right after I execute a Dial app. Can this be configured in extensions.conf? It seems the entries right after a Dial app get executed only if the Dial app was executed unsucessfully. Would I have to execute the dial app from the agi script? Thanks in advance. Dan -------------- next part

Hi all, I'm trying to build an IVR using the Perl AGI module at http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm But, I'm having trouble getting my program to play a message and wait for a keystroke. I am able to use this code to play the file, so I know that the $msg variable points to a valid sound file: $result = $agi->exec("background $msg");

Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

At the moment I'm prototyping an advanced ENUM application with PHP fetched from LDAP. When a user enters a full hostname as SIP adress I get loop problems from the AGI EXECUTE DIAL and from a Dial in the extension.conf. -- Executing AGI("SIP/1000-c3c3", "enum.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php enum.php: 123 enum.php:

Well guys here comes the fun part. I have a Microsoft access (VBA) application that interfaces with my SQL database. This app pulls of info from the SQL record and than picks up the phone and dials that locations number. I have purchased a few hundred NpaNxx's for my own use. I want get into too much detail there but no worries this is legal. I need to change my CID info on the fly. So I am

Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I