similar to: how to have an agi check for dial tone on analog lines before dialing

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten =>

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

The way I have my dialplan configured, an internal extension is routed to a different context (with Goto()) on pretty much the first button press. 2 -> internal extensions 0 -> operator 5 -> VM 9 -> outside line etc. So a "201" will go to the internal extensions context, s,1, do some setup and then match on "01". The thing is that when the 9 is entered, I

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

Hi List, I have this good looking IAXy device... I have managed to provision it, i can see it registering to my asterisk box, however when I pick up the phone which is plugged in the IAXy I have no dialtone, nothing. Any ideas what might be going on? Cheers, Jean-Michel.

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? How do I bring people into a meetme and play a message to all of them when they are on SIP channels? Jerry

At the moment I'm prototyping an advanced ENUM application with PHP fetched from LDAP. When a user enters a full hostname as SIP adress I get loop problems from the AGI EXECUTE DIAL and from a Dial in the extension.conf. -- Executing AGI("SIP/1000-c3c3", "enum.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php enum.php: 123 enum.php:

Hello List, This is my first message herein. I was playing around with System() and AGI() and found out something I cound not determine my configuration error. I added before.agi and after.agi to the agi-bin dir. Tried to make before.agi get run before the dial call and after.agi be run after. Only the first priority (step 1) gets executed. Here follows some relevant part of the tests: On

I have two possible senarios for making a call from an AGI.. Senario1 - Using GoTo In the extensions.conf I have.. [dial-out] exten => _9.,1,AGI(myagi) exten => _9.,2,Dial(SIP/blah/${EXTEN:1}) In the AGI I have.. EXEC GoTo dial-out|9555678|2 So using this method I don't have to really edit the AGI ever if I change the dialplan around as long at the context is correct.. At the end of

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Hi folks -- I have a FastAGI Perl script running, handling calls. It works great. At one point I have a Dial() command. If the called party hangs up, Dial() returns 0, and when I call my own recordCdr() function using the channel variables ANSWEREDTIME, DIALEDTIME and DIALSTATUS, everything is fine. However, if the called party picks up, and then the dialing party hangs up Dial() returns -1,

Hi guys, I'm writing an AGI program and use EXEC DIAL to do the dialing. The result reply from Asterisk doesn't come back until the dialing times out, or the channel is hung up after the call is connected with remote party and finally hung up. Is it possible to tell from the return code, or whaterver other ways, that the EXEC DIAL went through and the call was connected? Or that the

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

Hi, I have come across an odd problem. Basically I am transferring a call to an agent. The agent is logged in and set as paused. In order to find which agent to call I am using a fastagi script to just set a variable. When it falls through the agi script and dials the agent (using the variable) it doesn't connect the call properly to the agent. I get the beep but no audio (along with

Started a new thread as my problem is somewhat different than the OP. Seems his somewhat different problem doesn't work as advertised either. Eric Wieling wrote: > I don't know what the problem is, but this is what I use and it works on my analog FXO port. > exten => _9NXXNXXXXXX,1,Dial(${PSTN}/w${EXTEN:1}) So, I modified slightly to fit my dialplan: exten =>

Am using perl AGI to invoke the dial command thus: $AGI->exec('Dial',"$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002)"); What I expected that this will do is: 1. call the number using the string $numtodial2 - works OK 2. Set call limit to $maxcall and play a message $msgtime milliseconds before the call - works OK 3. On connect of the call send it to the macro conn

Hello * users, I know that passing variable in the AGI script is by exten => _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being passed and simple_c_prgm is C code Now how will I receive these variables within C code ? Is it by the same way arguments are passed in command line to C by using argc and argv or there is more to be done than that? Thanks Regards -- Arpit Mehta