similar to: Timing out issue whenusing AGI

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Hi all, I have the below peace of my AGI script...the problem here is that I cannot fetch the extension value to inside the script and assign it to another variable...I highlighted it in red #!/usr/bin/perl #use DBD::mysql; use DBI; use DBD::mysql; use Asterisk::AGI; ############################ #To read asterisk variable values. $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse();

I need some help with some statements..... #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); my $callerid = $input{'callerid'}; if ($optemp != 1) { my $empid = $AGI->get_data('employee',-1,5); $AGI->stream_file(entered); $AGI->say_digits($empid); my $optemp =

I've got a single phone line coming into an X100P. In extensions.conf I've got this: [inboundzap] exten => s,1,Answer exten => s,2,EAgi,hanguptest.agi I see the ring come in and Asterisk detects it and tries to do something with it: NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer("Zap/1-1", "") in

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

su testuser11 cd /storage/CME/test No problem. But when I try to access the same directory in windows I get these entries in my logs.... /var/log/samba/log.smbd ------------------ [2010/01/04 16:08:25, 1] smbd/sesssetup.c:reply_spnego_kerberos(350) Failed to verify incoming ticket with error NT_STATUS_LOGON_FAILURE! log.winbindd reports no errors so it seems that the SIU/UID mapping

I'm having trouble getting an AGI exec command to spawn app_disa. The script executes properly, but does not spawn DISA. The CLI gives no helpful clues. Am I doing the exec incorrectly? I want to have a way to authenticate callers to the extension by Caller ID... if their caller ID is in my database and set to active, they can call out. [like a calling card but auth'd by CID instead

Hi all, Is there a way to get the "called id" (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $callerid = $input{'callerid'}; $AGI->say_digits($callerid); } Thanks in advance, Angel.

I'm trying to dial using this script in perl: (asterisk_dial.pl) --- #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $pin = $AGI->get_data("beep", "10000", "3"); chomp($pin); $AGI->exec('SetVar',"NUMERO=$pin"); exit(0); --- This script sets 'NUMERO' with any value... In my

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

I've been trying to get a basic 5 question IVR survey working in an AGI script, and am having trouble with the SQL portion not updating the table. When I take out all the AGI references, and run just the perl script, the table updates with no problem(DBname,username,password have been substituted in this example for the actual values): #!/usr/bin/perl # # use DBI; $DATETIME =

Hey all, I'm sorry to bother you with something so trivial, but I seem to be having an issue with the Asterisk::AGI module. I am a relative newbie with Perl so it could be a stupid syntax mistake that I missed. It seems when I try to execute either the stream_file or the get_data subs nothing is actually done. It doesn't seem to stream the files, but on the console it says it played the

Hello. ???? ????????? ?????? ?? ?????, ??????? ??????, ??? ????????, ?????? ?? ???????????? ?????, ? ????? ???????? +?????????? ? ??????? ??????. I wrote a small perl script, that just calls to the specified number and then receives the information about +the status of the call. This script is below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input =

Hi all, Help needed: Does AGI SCRIPTS USING PERL NEED's SOME KIND OF COMPILATION TO WORK WITH *???? This simple script is not working. What can I do to make this interact with *????? #!/usr/bin/perl # # use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); my $tests = 0; my $pass = 0; my $fail = 0; #setup callback

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

I have a client who needs an application for there field techs to call in when they arrive on site and when they leave. The logic behind it seems pretty simple. I am going to write something in AGI to capture some DTMF tones and update this data into MySQL to run some reports from. But here's my initial problem. I have started to create a simple AGI script to capture the CallerID, but I