similar to: AMI Originate issue

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

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]

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

Dear all, I want to make call through socket i have set code given below: #!/usr/bin/perl -w use IO::Socket::INET; sub asterisk_command () { # my $command=$_[0]; my $ami=IO::Socket::INET->new(PeerAddr=>'127.0.0.1',PeerPort=>5038,Proto=>'tcp') or die "failed to connect to AMI!"; print $ami "Action: Login\r\nUsername:

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 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

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = "SIP\3846") Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The

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

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

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:

Hello , I have a problem to install asterisk with ldap. I am doing the following: make clean . / configure make menuselect LIBS =- lldap export LIBS make ====> This is where my error #make CC = "cc" CXX = "g + +" LD = "" AR = "" RANLIB = "" CFLAGS = "" make-C menuselect CONFIGURE_SILENT = "- silent" menuselect make [1]:

It appears that using dovecot to proxy to nginx imap proxy doesn't work. >From tcpdump and browsing the source it appears dovecot sends, C CAPABILITY\r\nL LOGIN <user> <pass>\r\n and nginx only responds to the CAPABILITY command. Is this a problem with dovecot sending the two commands without waiting for the first to complete or is it nginx's trouble with not handling it

hi I've written, or upgraded a little more, a plugin for asterisk/nagios, just in case someone should be interested. it uses the manager interface to connect and checks staus. it's a dirty hack, but it works. see https://sourceforge.net/tracker/? func=detail&aid=746083&group_id=29880&atid=541465 for more info roy

Hello All, My build fails in the "make asterisk" section, here is the log of the final steps it performed. All help appreciated... Thanks and Regards Ajit ============================================ make[1]: Leaving directory `/usr/src/asterisk/cdr' make[1]: Entering directory `/usr/src/asterisk/astman' rm -f *.o astman make[1]: Leaving directory

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 {