similar to: how to identify agi crash cause

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)

Just a shot in the dark... but have you tried Answer() before Playback()? Josh McAllister -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Engleward Sent: Monday, May 01, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup I have

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

Hi, For some reason, my AGI perl scripts cannot write to the CLI console using standard error. I ran the agi-test.agi test script that came with asterisk and verified that the problem was not with the code. Asterisk is always started with 4 or more v's, yet this the CLI output does not show up. Have there been any major changes to AGI in the cvs-head? I am using Asterisk CVS-D2005.08.05.

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Hi i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards their phone to a external number the CALLERIDNUM is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a

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

Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has

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'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel

Hi, I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2). I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I

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 Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091213/1f81f008/attachment.pgp

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 I am having some problems installing a paravirt RHEL guest on a deb amd64 xen machine uname -a = Linux debxen 2.6.20-1-xen-amd64 #1 SMP Tue Apr 24 22:23:22 UTC 2007 x86_64 GNU/Linux ============== xm info ============== host : debxen release : 2.6.20-1-xen-amd64 version : #1 SMP Tue Apr 24 22:23:22 UTC 2007 machine : x86_64

Hello, I've found a couple of previous posts on this subject, but with no posted resolution... I'm attempting to run * as a non-root user (asterisk), following the guidelines on the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root I can run * as my new user with "/usr/sbin/asterisk -vvvvc" without problem. However, I'm unable to run * using

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

Kurt, Cool idea and great "seeing new faces" on here proposing things on here and engaging with R-core on here. Some comments on the issue of fallbacks below. On Wed, Mar 27, 2019 at 10:33 PM Kurt Van Dijck < dev.kurt at vandijck-laurijssen.be> wrote: > Hey, > > In the meantime, I submitted a bug. Thanks for the assistence on that. > > > and I'm not