similar to: Calls authentication by IP address

I say this note is important only because I (a AGI PHP newbie) was tormented by this problem for many an hour, even though I'm sure it's documented somewhere or obvious to more experienced users. So as I was experimenting with AGI in PHP scripting I was baffled by why Asterisk was properly receiving AGI commands written to stdout but always returning "510 invalid command" to the

Hi all, Sorry for this elementary question (I'm a newbie). I'm trying to write an agi script (test.agi) and run it when I call in. However, I'm getting an error that says application agi isn't being found. I've put test.agi into agi-bin with permissions 755. Do I have to compile agi support into Asterisk, or is it built in? My test.agi script is php, but not using anything

Hello I'm pretty much a newbie when it comes to C, but I have to use this language to write a couple of AGI proggies because I need them to be statically compiled. Strangely enough, Google didn't return much when looking for the "Hello, world!" of AGI in C. The following doesn't work: The file never gets written: =========== //check_cid.c #include <stdio.h> #include

I'm trying set up asterisk on an amazon instance in Sydney. It's to use for our kids in Sydney to connect with their friends in the States. We've found iax works better than sip with these distances. But we now have weird problem: everybody has a cell phone, and it's much cheaper/better to use the house internet connection over the phones wifi. Each cell phone has it's own

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension

> > Hi every body, > > Please if some one could help me with this: > I'm writing an AGU Perl Script which basically makes a call using an extension provided by other asterisk box to an E1. The asterisk version is 1.6.0.28, so it hasn't the Wellington know AGI class. The code is as follows: > > ============================= > #!/usr/bin/perl > use strict; >

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Hi, I am a newbie and just discovered AGI (after learning a lot about extensions.conf's language). Before putting in a lot of time on AGI/Perl/PHP I would like to know if its possible to do most of the functionality performed in extensions.conf through AGI. Can AGI be used as a replacement for scripting a dialplan? Thanks, Michael

Hi Members, I was able to install Asterisk and configure many of it's features. Currently I am using Extensions.conf for giving all my contexts and extensions. Whenever I change my extensions or add a new context I have to reload extensions.conf and practically it is not possible reloading many times as we update or add contexts many times. Please tell me what could be the best solution to

i've just spent the pass 2 days trying to get AGI to work with PHP; i made a lot of silly mistakes along the way which could have been avoided if only there were some kinda howto or samples. at the risk of looking stupid, i decided to shared my experience in hopes that it might help some newbie get going with PHP. 1. first order of business is to be aware of your php environment; i m NOT

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

I've got an agi that recognises speech (via Google) and another that turns text into speech (tts) (via Microsoft Translate). Both are web APIs, both called via seperate python AGIs. I've googled and I'm probably missing something pretty newbie 101 here, but is there any way, or fiddle, that I can play some audio to let the caller know that their weather forecast is being fetched,

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? I'm a * and linux newbie, so please be gentle ;-) Thanks John

Hi, I am newbie to asterisk. I would like to use asterisk using VoIP. I don't want to use any hardware. I have installed Asterisk 1.2.13. I would like to record using AGI command RECORD FILE. I would also like to do conferencing and recording in asterisk. How could this be done? Help me out with conf files and also using AGI script. -------------- next part -------------- An HTML attachment

Summary: Is it possible to append to a user-defined text file from the dialplan without using AGI ? Background: We are using our Asterisk server to place outbound calls alone. We've 8 E1s that enables us to place about 240 calls concurrently. The server is a bit old, and we are not able to reach our maximum capacity. The dialplan makes use of MySQL to record call status information to our

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews & Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A&A were for some odd reason not sending calls to my server, but that has been resolved. The problem now is that the calls fail to

Hi, sorry if this is a newbie question, but in fact I am sort of a newbie. Is there a way of connecting two answered and active voice channels together in an AGI script for some time, having the two parties talk to each other, at the same time have asterisk or the AGI script listen for DTMF tones on both channels and react to certain tones, i.e. disconnecting the two channels on reception of