similar to: AGI and forking

I believe I am running an AGI (to put users in a conf) before the confbridge is built. So the users are not really get in the conf... exten X,1,run agi to put users in conf exten X,n,ConfBridge() How do I have in the dial plan ConfBridge() and someplace run an AGI that brings the users I want into that Conf. I cannot delay in the AGI and wait for the conf because the conf is not built until I

If anyone can help me with this I'll be soooo grateful :) We have a isdn30 line, with a DDI range. We also have 2 business units that have separate 0870 numbers that are mapped onto 2 DDI numbers. I would like to be able to present these 0870 numbers from the business units so that the correct number is displayed on a callerid, or when 1471 is dialled. BT claim that I can only have a

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.

Hi, The systrace and rlimit sandboxes have been committed and will be in snapshots dated 20110623 and later. This diff adds support for pre-auth privsep sandboxing using the OS X sandbox_init(3) service. It's a bit disappointing that the OS X developers chose such as namespace-polluting header and function names "sandbox.h", "sandbox_init()", etc. It already forced me to

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

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

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

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

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

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

Hi, This patch (relative to -HEAD) defines an API to allow sandboxing of the pre-auth privsep child and a couple of sandbox implementations. The idea here is to heavily restrict what the network-face pre-auth process can do. This was the original intent behind dropping to a dedicated uid and chrooting to an empty directory, but even this still allows a compromised slave process to make new

I have an ITSP we are trying to work with that has an "Unusual" way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: <sip:s at x.x.x.x> This then causes ALL

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

Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

I have asterisk 1.2.18 running on a new system we just installed. Although I've used AGIs many times in the past, I'm stumped on this one. It may just be a simple issue that I need another eyeset to look at. My AGI does the following: #!/usr/bin/perl #Load a few modules... use Asterisk::AGI; use DBI; $AGI = new Asterisk::AGI; #Grab input from Asterisk my %input =

Hi there! Samba-2.2.8a There is possible deadlock in smbmount (smbmount.c) when parent process forking. Maybe this is a Linux-only bug (2.4 and 2.6 shows the same). But this is enough to apply my patch (or make other proper thing[s]). Explaining: Smbmount need to be a daemon to serve the mountpoint. Parent make [sys_]fork() to create this young daemon. After sys_fork() children go ahead and