similar to: How to run a remote PHP script and still have access to audio stream?

Hello everyone, I'm having all sorts of problems with my PHP AGI scripts... Basically, my scripts run fine from the command line and don't do anything well called from Asterisk. Here are my questions: a) Does Asterisk require PHP CLI or CGI? From the command line, my script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi) b) How to debug my script? According

Dear Moises Silva, Chris Thompson and others, I've finally managed to find the problem with my PHP AGI script! Apparently, my machine has 2 versions of PHP installed. One for CGI and the other for CLI. Without knowing, to hand test my script I was calling the CLI version (/usr/local/bin/php), whereas Asterisk was calling the CGI version (/usr/bin/php). It turns out that my script

Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

Hello there, Somehow, Asterisk log files are consuming all the space that I have in my hard disk... They've already eaten 14GB and are still hungry!! What shall I do? I'm not even logging anything in verbose mode!! Help really appreciated!! Best, Leo

Hello there, I'm new to Asterisk and I'm trying to figure out a way to make the Asterisk Manager Interface (AMI) accessible to multiple users at the same time. Would anyone recommend an AMI proxy that could be accessed from PHP code? Thanks in advance, Leo

Hello everyone, Where can I find instructions on how to install PHPAGI? BTW, what's the difference between PHPAGI and PHPAGI2? Are they different products? It's hard to tell from voip-info.org... Best, Leo

Hello there, I'm a new Asterisk user and I'm having difficulties to connect to and from my Asterisk server. Can anybody give me a hand? Here's some background information: * I'm running RedHat Linux Enterprise 4.0 * When iptables is stopped, my server can register with IAX service providers and receive registrations from IAX softphones. However, it does not succeed in

hi all, We going to implement a music service which enable user to playback a song by dialing to a service number. The problem is that the amount of data is huge so we have to plae it on an different server which is connected to the asterisk's via internet. Does asterisk support playing a audio file from an resource locate in a remote host? Please help, Thanks, Quyps

Hello there, I'm new to PHP AGIs and I'm having problems with a particular script that has a "include_once" statement on it. If I remove that stament, the script runs until the section of the code that depends on the include and then returns. If I include that statement, the script does not seem to run at all. What shall I do? Thanks in advance, Leo

Ok, here's a weird one. I have an AGI script where one user calls another. The call is answered. Everything is peachy. If the call is terminated by the CALLEE hanging up the call, then Asterisk returns control back to where the Dial() command left off, and I can check the return code of Dial(), ${DIALSTATUS} etc. That's all great. HOWEVER, if the CALLER hangs up the call, it seems as if

Hello, I have problem with creating user under centos: part of my class: user {jboss: ensure => present, home => "$jboss_home", shell => "/bin/bash", groups => jboss, } and in log I see: err change from absent to present failed: Could not create user jboss: Execution of ''/usr/sbin/useradd -d

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

hi all, I wish to draw on the same graphic device 3 functions. But i don't want them to be on different graph, i want to compare them on the same I don't need mfrow or mfcol, I need something else... 1 graph on 1 device inside this graph 3 ploted function. I saw something unsing data.frame, but i think it's overkill, and something less complicated must exist, if not why? why not

Hello all, I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part

Hi, I'd like to know if there's an "easy way" of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like "monitor" except audio should be sent live. More like "app_ices" (or

Thks for your answer, here is an exemple of what i do with the errors in french... > tmp [1] 200 150 245 125 134 345 320 450 678 > beta18 Erreur : Objet "beta18" not found //NORMAL just to show it > eta [1] 500 > func1<-function(beta18) dweibull(tmp[1],beta18,eta) > func1<-func1(beta18) * function(beta18) dweibull(tmp[2],beta18,eta) Erreur dans dweibull(tmp[1],

Greetings all- I'd like to install Oreka on a Centos 5.x server for monitoring my Asterisk systems(using port mirroring) but I find I'm having problems with the version of libpcap installed. The latest is 0.9.4 but the orkaudio RPM (built for Centos 4.2) requires libpcap 0.8.3. I've tried making symlinks to overcome the issue but with no success. Without having to build Oreka from

Hello there, I'm writing an application to display asterisk voicemail on a webpage. Since Flash only handles MP3 files, I wonder if it is possible to configure asterisk voicemail in such a way that it would record and play MP3 files... Would anyone give me suggestions on how to do this? Is format_mp3 stable enough for something like this? Thanks in advance, Leo Burd