similar to: perl --> manager problem

Hey Guys! I have perl script for allpage which is working fine with asterisk 1.8.2.3 version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is there anything changes ? If i run this script from command like it works but not from asterisk dialplan. This script nothing but just connecting AMI interface and using Variable: SIPADDHEADER=Alert-Info: Ring Answer variable to

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 Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:

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 Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

Hi, I have written a C code which would let me connect to the Asterisk Manager Interface. The code compiles successfully but on running the code I get unauthorized login shown in the Asterisk command line console. Here is my C code: #include<stdio.h> #include<netdb.h> #include<unistd.h> #include<string.h> #include<arpa/inet.h> #include<sys/types.h>

Hi, I'm trying to write a simple perl script that will run the following: Action: Originate Channel: local/xxx@callback/r/n Exten: 1234 Context: callback Priority: 1 Extensions.conf exten => 500,1,agi,callback.pl callback perl script: use Net::Telnet (); $mgrUSERNAME='fred'; $mgrSECRET='bloggs'; $server_ip='127.0.0.1'; $tn->print("Action:

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who

I'm trying to use accountcodes, but experiencing inconsistant results. I have two * servers, one which appears to be working as expected and one not. I would like to prepend the device's accountcode to the dialed number. The sip1 server does not seem to have the ${ACCOUNTCODE} variable set when reading the extensions.conf, but sip2 server does. What troubleshooting or trace

I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of

Hello, I'm trying to investigate and solve some postgres latency spikes that I'm seeing as a result of some behaviour in the syncer. This is with FreeBSD 8.2 (with some local modifications and backports, r231160 in particular). The system has an LSI 9261-8i RAID controller (backed by mfi(4)) and the database and WALs are on separate volumes, a RAID 6 and a RAID 1 respectively. It has

https://bugs.freedesktop.org/show_bug.cgi?id=98030 Bug ID: 98030 Summary: Stuttering video playback in totem after update to 1.19-rc1 Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Hi, First things first, sorry for the off-topic, but I've already burned my eyes Googling and couldn't find the answer to what I need and I remembered asking here, because I'm a long time CentOS and its mailing list user. Again, my apologies. I have this command to create an FTP account: # pure-pw useradd mario -u 502 -g 502 -n 1000 -N 200 -d /home/pages/mario This command asks for

List: This is my first manager client that I've written so please bear with me: I am trying to write a C manager interface client to interface with our CRM software. I am having an issue while reading the data from the manager interface. I am writing this in C and I have the following code: while(1) { bzero(buffer,sizeof(buffer)); readCode = read(socketHandle,buffer,sizeof(buffer));

Is there a way to remote reboot a Cisco 7940 or 7960 phone via some kind of command? The idea is to force a reboot automatically after changing one of the configuration files. Regards, Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070527/857bdda8/attachment.htm

## Not run: from <- sprintf("<sendmailR@ to <- "<olafm at datensplitter.net>" subject <- "Hello from R" msg <- "It works!" sendmail(from, to, subject, msg, control=list(smtpServer="ASPMX.L.GOOGLE.COM")) ## End(Not run) the above commands are provided in this document ie http://cran.r-project.org/web/packages/sendmailR/sendmailR.pdf

I've been seeing some random Proliant DL380 G4 64bit crashes. Each time, on the console are messages relating tojbd2/cciss and something about a waitfor 120 seconds. Is anybody else seeing anything like this? Oddly, I can't seem to find this in the logs. I guess it can't write when this happens. -- John Hinton 877-777-1407 ext 502 http://www.ew3d.com Comprehensive Online

I built asterisk-trunk at 92526 and asterisk-addons-trunk at 496. I have my Bluetooth cell phone connected. It almost works. In mobile.conf, I have "context=incoming-mobile" for the phone, and that looks like: context incoming-mobile { _. => { VoiceMail(9999,b); Hangup(); }; } Calls to the cell phone get directed answered by Asterisk and directed to

Hello. I am writing a back end in which I combined the existing BPF LLVM back end with the Mips MSA vector extensions (from the Mips back end) I have encountered an error when compiling with llc: the instruction selector uses a vector register instead of a scalar register with type i64 . I have the following part of LLVM IR program: vector.body.preheader: