similar to: how to debug perl agi

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:

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

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

hello like if 6000 is the main exchange number. any one dial to 6000 will be asked for pressing his desired extension then he can press his desired extension then his number is diled exten=>6000,1,Background(enterdesiredexten) exten=>6000,2,Wait(2) exten=>2000,1,Dial(SIP/${EXTEN})

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 have problem in compiling asterisk-addons 1.0.1 --------------------------------------------------------- [root@kamran asterisk-addons-1.0.1]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c ../asterisk: Not a directory make: *** [cdr_addon_mysql.o] Error 1 --------------------------------------------------------- i want to

hello i was searching for solution to problem (sip->h.323). any one from this list asterisk mailing have any idea how to fix it. i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do

hello i am trying to make a callback solution. client will call callback number and call is terminated. now callback server will create a call for that client. actually i have a problem in this process. that server is creating call to client (UA) when previous call is not disconnected yet. UA---------->Asterisk(callbacknumber) callis answered UA<----------Asterisk(callbackserver) call is

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

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 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:

hello all i am having a problem in compiling openh323. [root@kamran openh323]# ./configure checking for g++... g++ checking for C++ compiler default output... a.out checking whether the C++ compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C++ compiler... yes

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Hi, Microsoft Dynamics CRM 3.0 integration with Asterisk/Trixbox has been included in StarJunction and Star Outlook Dialer. This is in addition to existing support for SugarCRM and Salesforce CRM. It is available at http://www.starutilities.com/staroldialer.htm Thank you for your valuable comments and suggestions. Kamran

i am not able to communicate with ip scheme 172.16.x.x but when i change it to 192.168.x.x it works properly any one help me __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Hi Everyone, New to the list. Thanks in advance for any assistance you are able to provide. I have a TrippLite SMART2200RM2UN UPS. I have installed and configured NUT as instructed on the website, and am able to monitor the status of the UPS without much problem. The only problem I am seeing is that I cannot get the machine to actually send a Low Battery ( LB ) signal. When I run

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

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

Hello everyone: I have a problem with the shorewall configuration. Let''s me tell you. I have installed shorewall 2.0.4 into a machine with 2.6.8 kernel. This machine works like a software-router: it has 2 netcard eth0 goes to the local network 192.168.0.0/24 eth1 is an interface for ppp0 (there is an ADSL conected) I have defined the Network Zones (net, loc); The Network Interfaces

Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf