similar to: Asterisk and Databases

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

hi everyone, actually i want to release an IVR system using PHPAGI API , in this IVR i want to get value from the user. I already used get_data defined in phpagi but they are not able to get the value given by the user and store it in a php variable. i tested this : $result = $agi->get_data('beep', 3000, 20); $keys = $result['result']; but every time i found in $keys variable

> I'm going blind searching - maybe you know? > > During the execution of a script I want to play fake ring to caller. > Both of these examples complain of missing option: > > $agi->exec("Ringing"); > $agi->exec("Playtones ring"); > > Notice: Undefined variable: options in > /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

exten => auth_dial,1,DigitTimeout,5 exten => auth_dial,2,ResponseTimeout,15 exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0) exten => auth_dial,4,agi(call_start.php|${dialed}) exten => auth_dial,5,dial(SIP/${dialed}@146.82.15.241) I'm trying to get What they dialed put into the PHP script. How do I get the contents of this variable in the php script?

Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some

Hi All... I have a need to phone a large number of people and collect information from them. I know Asterisk has a nice IVR system, but can it be used to initiate a call to people listed in a database or text file? Don't worry, this is not an annoying marketing thing. Thanks...

Hello, I'm trying to understand the AMI and use the PHPAGI library to extend and to easy my work ... I'm still trying to figure out how things works, but I haven't found enough resources to make me sure about certain things ... So, I'd like to know if somebody can give me a hand .... ;) My calls are answered almost instantly by an IVR and puts the call on a queue to wait for

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Hi, We have a "simple" AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect

Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|speex), > priority = mine -- Executing

Hi guys, I hope I get your help for this problem ... at least point me through the right direction .... I configured a queue that when the people are tired of waiting they can press 1 and I direct them to voicemail ... but I'd like to make a callback solution at this point, because during busy hours it is hard to check the voicemail ... I'd like to make my system to call back this

Hello guys, Does anyone has an Asterisk server hosted off-site ? Like in those data centers that do web hosting in dedicated servers ? Is there a hosting company that has a special plan to host voip services like this, or usually is hosted in those dedicated servers like the ones I asked above ? What about QoS ? I know that most (if not all) are connected to their switch through a

Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]

I have the following code. When I call the extension, it either ignores the first "Hello there everyone", or says "hello" and moves on sometime stoping before it finishes hello. The rest of the text reads fine. Anyone else have this issue?? Thanks! require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi->answer();

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",

I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go

I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? The backgrounder behind my question is