similar to: Voice analytics

I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -------------- next part -------------- An HTML attachment was scrubbed... URL:

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

Hello, I'm using Hanhup Handlers in a testing asterisk 11 system. Within one such handler, I'm setting CDR values. To me, it seems those changed CDR values are not saved in CDR back-end. Can you confirm ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

Julian, I did not understand what you meant by this paragraph. "Now, for the twist. For development and testing, I assigned an ip address and gateway to the bridge. I need to be able for a "non-it" person to install this box without having to set it up at all , so it cannot have an ip address assigned, as it *may* be in use somewhere else on the lan or router." Did your

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

need transcoding to a|ulaw. I am using it with no problems (have g729 licenses as well though). A bit off topic, I have found some extra configuration that is not really in the docs (or I could not find them): fullname=Your full name country=gr language=en city=City province=Province phone_home=+fullinternationalnumber phone_office=+fullinternationalnumber email=your at email.com

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

Hello, I am trying to run a small application called Miku Desktop Mascot. a small anim character dancing on the screen (nice distraction when all day on code lines). current system is : Fedora 13 x64 Nvidia GTX 460 , proprietary driver 260.19.12 Wine 1.3.5 x64 I actually am a java developer (not c++) so if I can be of any help testing, compiling wine on my machine it's possible. thanks

Full_Name: Lyndon Drake Version: Version 0.65.0 Under development (unstable) (August 07, 1999) OS: Linux 2.2.2 Submission from: (NULL) (202.14.102.52) I was attempting to install the DSE package from a directory whose pathname includes spaces. The INSTALL script interprets spaces in the pathname as a separator. For example: [lyndon@beat Rdse]$ pwd /home/lyndon/doc/uni/415.704 Computer Systems

Hello, i am trying to proxy a LMTP connection with version 2.0b3 Currently i have the problem when trying to use a named based proxy for LMTP the process doesn't resolve the hostname and crashes: Feb 26 16:53:26 auth: Debug: ldap(vodafonemail56 at vodafone.de,::1): pass search: base=ou=mailboxes,ou=vfag,c=de,o=vodafone scope=subtree filter=(&(objectClass=uco)(mail=vodafonemail56 at

Well, it looks like AstriCon 2006 is going to be big. We've sold out the entire Westin Park Central -- every last room. So, here are some nearby hotels to check if you're planning on coming down to Dallas for the big Asterisk-fest. Wyndham Garden Hotel-Park Central 8051 Lyndon B Johnson Fwy Dallas, TX 972-680-3000 Residence Inn LBJ/Park Central 7642 Lyndon B Johnson Fwy Dallas,

* * * * *Senior Statistician* * * *Company Description*: Health Market Science is a young, rapidly-expanding company located in King of Prussia, PA. HMS is the leading provider of scientifically-engineered targeting, data integration, and advanced analytics solutions for the healthcare industry - Finalist for the Technology CEO of the Year in the 2009 Enterprise Awards Program -

I'm using FreeNAS 9.1, and I've just got an APC Back-UPS CS 500. In theory it's supported by NUT, but the latest released version (2.6.5) doesn't have support for it (the driver reports an error, which is apparently due to an overflow). According to a thread on the mailing list, Back-UPS ES series requires a fix that's in source control, and CS series requires an additional

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

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

I''m doing a graphics application and need onscreen anti-aliasing for both lines and text. Seems like wxGraphicsContext would do that, is that planned to be ported to wxRuby? Best regards, Jari Williamsson