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

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

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,

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

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");

I posted earlier in the day about needed help with IAX trunking. I did some more reading and made some more changes. Here is what I have thus far: Iax.conf on one server [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [vvfarm] type=friend username=colo secret=testpassword auth=plaintext host=64.194.211.170 context=iax-incoming

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

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

Okay, so my issue isn't really a technical one but more of needing advice on the best way to program this. I have a user in Colorado who works from home but frequents our office in Colorado. All of our remote users connect to a server in Dallas the users at the HQ in Colorado connect to a separate server in the Colorado office that is on the private network. I have an IAX trunk between the

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

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.