similar to: how to connect to asterisk via perl agi

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

Dear all, I have installed asterisk 1.0.5 on redhat 9 I have installed also, asterisk-oh323 0.6.5 module (successfully compiled and installed) Now When I am trying to get asterisk communicate with a Radius (in my case: it's the VoiceMaster Radius) I was able to do the following: After installing all recommended to download and install radius client for asterisk

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

hello i am using ast-rad-acc.pl from portaone connected with asterisk manager. my (%cdr) = @_; $cdr{'CALLERID'}, $cdr{'DNID'}, these are empty why these two variables are not working on hangup any comments thanks Kamran Ahamd __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

hi any one tell me how to make a dialplan my extensions.conf exten => _40XXXXXXXXXXXX,1,Dial(OH323/${EXTEN}) i want to dial to 40XXXXXXXXXXXX number. XXXXXXXXXXXX could be any number like 923335224005 or 92512213248 at the moment when i am trying to dial 40923335224005 asterisk is dialing Executing Dial("OH323/R11429", "OH323/40923335224005") but i want him to dial

hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten => _X.,1,agi,agi-rad-auth.pl|Routing=SIP&AuthorizeBy=SIP vi /etc/asterisk/modules.conf load => res_agi.so <---------------errors------------------------> *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains:

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

hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason regrads Kamran __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on

i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. ------------------------------------------------------ *CLI> odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI> ------------------------------------------------------ and user is added to

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 ...

Hello, As part of the Rust bindings for Libnbd, I try to integrate the asynchronous (aio_*) functions with Tokio <https://docs.rs/tokio/latest/tokio/>, the most used asynchronous runtime in Rust. However, in its eventloop, Tokio uses epoll(7) instead of poll(2) (which is used internally in Libnbd). The difference is that poll(2) uses level-triggered notifications as aposed to epoll(7)

I here is some information I could have provided previously. Here is what I am seeing: ==================================================================== Broadcast message from nut at localhost.localdomain (Fri May 13 22:31:35 2016): UPS myups at localhost on battery upsc myups at localhost battery.charge: 3 battery.charge.low: 10 battery.charge.warning: 30 battery.runtime: 93

Hi Charles, I made the change, and it still won't incite a poweroff: ========================================================================== [root at localhost ~]# upsc myups at localhost battery.charge: 76 battery.charge.low: 90 battery.charge.warning: 30 battery.runtime: 1811 battery.temperature: 31.9 battery.type: PbAC battery.voltage: 49.2 battery.voltage.nominal: 48.0 device.mfr:

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 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]:

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 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

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 experts, to avoid any unknown problems with my Linux installation I have now as a last resort method installed SuSE Linux 9.0 a new and have downloaded a fresh copy of Asterisk via CVS. Then I followed the steps of the "Getting started with Asterisk" and compiled successfully zaptel and libpri (as far as I can see). But when I compile asterisk I get an error. I have attached the

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: