similar to: Manager ExtensionState function

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

hello i am getting this error while trying to run ast-rad-acc.pl my $ast_connected = 1; while( 1 ) { if( $astman->connect ) { $ast_connected = 1; syslog('info', 'Connected to Asterisk!'); $astman->setcallback('DEFAULT', \&status_callback); eval { $astman->eventloop; }; print STDERR "Connected To Asterisk\n"; } else {

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available" back from xxx.xxx.xxx.xxx:5060 [Mar

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

Hi, I am looking for a simple solution to do this. I wish to have the user to enter their preferred method of connection i.e. for the cheapest solution to their desktop phone or mobile phone, then plan callfile based on the number that user provided and dial to the user. Any suggestions? CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

Hello ppl, Have implemented a really nice Billing engine using AGI scripts. So far it works fine, tho haven't yet put it in the torture cell. The AGI scripts have been written in PHP, using MySQL for the billing and profile information. The major disadvantages I see using AGI scripts : 1. A new process(invocation of PHP scripts) on every new call. 2. MySQL connections on every instance of

I am wondering if anyone can help me figure out how to do something. I'm running a simple perl AGI which at the end of the call, creates a MySQL row with some of the tasks done during the AGI session. However, when the call is unexpectedly dropped (possibly the caller hangs up before finishing the entire AGI flow), I don't get the MySQL row created. I'm wondering if there is

I have an asterisk servicer (1.0.5) with 3 X100P cards. Everything is working fine but two days ago I implemented call forwarding using the example from voip-info wiki. Now when I enable call forwarding on my phone and a call comes in it gets redirected to my cell and everything is apparently working. The problem is that when we hang up both Zap interfaces (the one where the original

has any body considered using astman/gastman to show a) sip/iax etc registry status with other * b) manage multiple * boxes would this be any use, i was just thinking if some tech support has to handle multiple * for one organization, it may be worthwhile to have mutiple * boxen become part of one REALM and manage that realm, or maybe some other way, but i'm sure we can find someone to

Signed-off-by: Joey Boggs <jboggs at redhat.com> --- scripts/ovirt-config-setup.py | 42 +++++++++++++++++++++++++++++++++++----- 1 files changed, 36 insertions(+), 6 deletions(-) diff --git a/scripts/ovirt-config-setup.py b/scripts/ovirt-config-setup.py index d55ea36..753c80d 100755 --- a/scripts/ovirt-config-setup.py +++ b/scripts/ovirt-config-setup.py @@ -231,7 +231,8 @@ class

On Solaris 8, I have ssh 3.1.0 and on other box Sol 7 I have 1.2.26 (min version for comtable with ssh 3), I checked also /etc/ssh2/sshd2_config file ## SSH1 compatibility # Ssh1Compatibility <set by configure by default> # Sshd1Path <set by configure by default 2) generate key for ssh3 # ssh-keygen2 -P /etc/ssh2/hostkey

New installation from AsteriskNow 3.0.0 with asterisk 11 and freepbx, running Digium D40 and D70 phones. Direct-dialed extensions work fine, but extensions in RingGroup won't ring - dialparties.agi apparently removes them from the dialstring pre-emptively: dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN) dialparties.agi: Extension 2010 has ExtensionState: 4 What can I do? Any extension

Hello List, I have been working on a PHP application in order to build a BLF style script. Until now everything is going Ok but something a little (in my oppinion) strange is going on with the 'ExtensionState' command; The problem is that it does not returns the 'Status' as it's suposed to, mentioned in the A.T.F.O.T book - version 2., where it sais something like:

I'm not having any luck getting the ExtensionState action of the Manager API to work. The response is always success but the Status is always -1 which to me means an error. Here is a typical telnet session. ----------------------------------- stockholm:~ # telnet localhost 5038 Trying ::1... telnet: connect to address ::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape

This is my first attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then when it has the answer it outputs an XML file for use as the directory on a Cisco 7960 phone. What I am thinking is that when the user hits the directory button to veiw the directory that is at this URL it will