similar to: Really Big Queues

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

Hey, Is there any way to share?MySQL?connection between different agi's.Actually when call comes to asterisk box it executes various agi scripts sequentially. Each script checks various values by making a new?MySQL?connection and then execute query and then disconnects.? So,?Ideally?there should be one connection, and it should be reused between each agi and when a call is over it should be

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

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Thanks for your quick replies - I had no idea this list was so responsive! One more issue for you - we are trying to build a custom voicemail application and so far everything works great by using AGI to play prompts, get input and record files. However one of our customers has asked for some pretty advanced functionality, specifically the ability to: * stop in the middle of recording (so far so

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

Hi Everyone, I'm wondering if anyone here write AGI's in compiled binaries. I'm writing a small Cepstral AGI in Freepascal/Lazarus. I know there are some other AGI's out there, but I wanted to add some more functionality than what is available such as having the AGI determine if the "data" argument is plain text or a path to a text file and act accordingly. The

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

I have a SIP phone calling an AGI application. It starts out this way: -- Executing [s at macro-Call-AGI:2] AGI("SIP/151-b414f0c8", "computer-temp.sh,darwin,") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame

Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to

AGI's are resulting in unusual behaviors. Can someone please tell me if this is my inappropriate use of AGI's, inappropriate use of Time::HiRes, or a bug with *: I call this script twice: #!/usr/bin/perl use Time::HiRes qw( gettimeofday ); ($seconds, $microseconds) = gettimeofday; $hirestime = sprintf("%s","$seconds$microseconds"); print "SET VARIABLE

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

Is there an implementation of the Cox-Snell residuals / Nelson-Aalen plot for goodness of fit? Or otherwise is there an appropriate Goodness of Fit diagnostic? Thanks Murray -- Murray Pung Statistician, Datapharm Australia Pty Ltd 0404 273 283 [[alternative HTML version deleted]]

Asterisk Users In an attempt to help Asterisk move forward, a number of us have decided to create a book. It would initially be released as an "ebook" that could be sent to newbies to help them up the rather steep learning curve. Ultimately I would like to see it published and sold in bookstores (preferably by O'Reilly & Co.). Below is the outline for the book. We REALLY

List, I finally decided to break down & start playing with AGI scripts, but for the life of me, I can't figure out what I am doing wrong. Below is a super simple script to run a query in mysql to see how many call records there are for the extension calling in, then print the total in the CLI. This is all I get on the CLI: -- Executing AGI("SIP/216-0baa",

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

Hi, we are experiencing a strange issue and I am hoping someone can point me to the right direction or help out with some pointers. We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4 T1's for a total of DAHDI 96 channels. We have an agi application (php) that acts as a kind of a calling card application. All users are SIP users that make calls and asterisk then bridges

Hello list!! I want to connect an open call with an extension. I call in with a DID, them redirect to the extension using AGI. Can I use agi's originate to make the second call without dropping the first DID call? How would I go about this? I had something like this in mind: first answer the DID call, then with AGI: Action: login Username: xxxx Secret: xxxx Events: off Action: Originate

Hello all, I've seen that the application AgentCallbackLogin()has been set to deprecated in version 1.4. So I've done some tests based on the tutorial "queues-with-callback-members.txt" coming with version 1.4. What's not clear for me is what is happening to agents.conf, it seems that it's no longer needed, and I have to define my agents using variables in

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I