similar to: AGI Module

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

I wanted to post the beginings of my latest IVR Project for an automated Time Clock software. The customer has over 300 Field Reps that call in everytime they arrive on location and whey they leave that location. This is handled by the receptionist now and she logs in them and out of there Time Clock Software. Which takes up majority of her day. The customer has requested a automated way of

Hi Asterisk-Users, We have a problem with queues.conf / extensions.conf queues.conf file reads like ... member => SIP/8399 extensions.conf reads like ... exten => 8399, 1, SetCIDNum(${AccountNumber}|a) exten => 8399, 2, Dial(SIP/8399,10,Ttrf) When somebody calls to the queue, we observed that it is not going through extensions.conf (previous two lines) That mean's it is not

Hi All, When I call the stream_file function all goes well if the user doesn't clear the call. But if I do clear the call (on the handset for example), I get the following exception: -- Channel 0/31, span 1 got hangup RESULT_LINE: 200 result=-1 endpos=28000 == Spawn extension (default, 600006, 1) exited non-zero on 'Zap/31-1'

Hello, I want to create php rate script and I'm using Deadagi. But I allways get billsec 0 , or nothing. Can you help me to solve this problem... My extension.conf: exten => _123,1,DeadAgi(rate.php) exten => _123,2,hangup And my simple test php script rate.php #!/usr/local/bin/php -q <?php include_once (dirname(__FILE__)."/phpagi.php"); $AGI = new AGI();

I need some help with some statements..... #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); my $callerid = $input{'callerid'}; if ($optemp != 1) { my $empid = $AGI->get_data('employee',-1,5); $AGI->stream_file(entered); $AGI->say_digits($empid); my $optemp =

Hello Users, I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons -1.2.4 in My Linux. For My Voice messages Storage , I want To Use the MySql. In Googled it shows me the ODBC integration.. Is it need for that ODBC integration with MySql for my Voice Message storing in MySql. When I'm trying to integrate with ODBC +

I've been trying to get a basic 5 question IVR survey working in an AGI script, and am having trouble with the SQL portion not updating the table. When I take out all the AGI references, and run just the perl script, the table updates with no problem(DBname,username,password have been substituted in this example for the actual values): #!/usr/bin/perl # # use DBI; $DATETIME =

Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

Im trying to create a custom application via the AGI. I want to authenticate the users that dial in with a userid and pin. However, the number of digits in the PIN and userid are variable, and therefore I need to allow the user to "press enter" by hitting the pound key. How would I accomplish this in the AGI? stream_file doesnt seem to work, since it only allows one digit to be

Is this a bug: Trying to update when the where condition gives zero rows throws an error on MS SQL server > sqlQuery(pipe,"select * from ComDetailCurrent where RateTypeId is null;") [1] ProcessDate SourceSystemId AccountNumber Xref1 <0 rows> (or 0-length row.names) sqlQuery(pipe,"update ComDetailCurrent set RateTypeId=1 where RateTypeId is

Is there a variable that can be used with the playback command to play all gsm sound files in a directory? For example like '*.*' <- which I know doesn't work. Is the only solution for this to create an AGI script and use the stream_file command ? Thanks, J.C.

Hi all, I'm trying to build an IVR using the Perl AGI module at http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm But, I'm having trouble getting my program to play a message and wait for a keystroke. I am able to use this code to play the file, so I know that the $msg variable points to a valid sound file: $result = $agi->exec("background $msg");

Hi all, Directory lookup, Asterisk@home 2.7, are this small bugs? case DIR_FIRST: $intro = ($operator ? "dir-intro-fn-oper" : "dir-intro-fn"); break; case DIR_BOTH: $intro = ($operator ? "dir-intro-fnln-oper" : "dir-intro-fnln"); break; case DIR_LAST: default: $intro = ($operator ? "dir-intro-oper" :

I am having a problem with "record_file" working properly depending on when it is called -- basically if it is called immediately upon a call, it acts like it does not hear anything from the callers phone (yes, my phone is setup properly and functions fine otherwise)... if I do a "background" or "festival" command before calling it, it works fine. Details below:

I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,