similar to: How to fail an AGI

I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

After testing some AGI's, I noticed several extra Asterisk processes. They are not zombies, but can't be killed by safe_asterisk. Nor will they die when CLI issues stop now. Then I read that each AGI spawns a separate Asterisk process. But all my AGI calls have apparently completed successfully. So there should be no reason for them to hang there. Several questions: 1) Under

Good afternoon list. I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi. But after running the script, it just returns me 0 (true). Thus: -- <SIP/213-00000019>AGI Script check.agi completed, returning 0 I tried putting the lines "return false;" or "return 1;" but did not change anything. Does anyone have a clue? Thanks, Rodrigo Lang.

Hi, Tomorrow, Friday Nov 16th, 2007 at 12:30 PM, we'll be exploring a simple, well-commented example of an AGI script for asterisk. I have absolutely nothing against GUI, but if you want to unleash the real power of asterisk, you'll need to get into AGI (or pay someone else to do it). Because asterisk solutions are mostly limited to your imagination (and a hired hand may or may not have

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks

Hi all - Searching for java agi in the mailing list archives turns up ancient posts. Anyone else using java for their AGI? How well is it working & what are you using? My script is pretty simple, and I could write it with perl easy enough, but I just would feel better if I can keep most programming code for our system in java. Thank you- Matt

Hi, Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter what my script returns (0 or -1), AGISTATUS always appears to be 0 = SUCCESS. I was wanting my script to be able to return a value to the dialplan and then test AGISTATUS but it looks like I'm going down the wrong path. Any suggestions? Thanks, Brian -------------- next part -------------- An HTML attachment

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->answer(); my $i; $i = $AGI->channel_status(); $AGI->say_digits($i); $i =

Hi All, sorry if I post again this e-mail but I think the first one was lost. I don't know if this is OT but I'm working in my spare time at a small hardware project that match to what's requested below. It's a board with Input/Output capabilities and 10Mbps ethernet interface. It has Microchip software TCP/IP stack on it. Being at a very beginning stage, you can see a little

Hello. Happy New Year to everyone. I have a small WISP and would like to have customers to call our number to check their balance. I am planning on writing an AGI with php so it can get the customer info from the customer database. I don't know how to interact with the caller while in the agi script so this is what I have in mind: [test-agi] exten => 33,1,Answer() exten =>

Morning All, Has anyone here successfully implemented skills based routing within queues? The concept behind skills based routing is fairly straight forward, and I know I could do it with multiple queues, agent penalties and a bit of AGI to put the call into the right queue. However doing this is going to require the addition of several extra queues and isn't a very clean solution. The

I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a

Hi All, I am trying to implement ACD using Asterisk 1.2.18 and I've chosen AgentCallbackLogin for login purpose. One AGI is written which will actually get executed when agent dials '1001' (say) from his SIP phone and enters into the queue. Second AGI gets executed when the Dial operation is performed. I see the agi_uniqueid obtained from both AGI instances are different and I

Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports "Web interface for

Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas?