similar to: AstriCon 2013 (our 10th AstriCon) needs YOU!

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13

I am looking for any information of the possible use of Asterisk in any of the systems used by/for the London Olympics. Please get in contact if you know of any such use(s). Thank you, David Digium logo David Duffett Digium, Inc.??? Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green ??? Bicester, Oxfordshire OX25 3SX ??? UK direct/fax: ? +1 256 428 6119 ??

It's amazing what you can learn in a few days... Having just found out that Queen Elizabeth has a great sense of humor, it has now emerged that Kevin Fleming - a man who (both with and without his moustache) has been an amazing contributor and influencer in the Asterisk project is set to move on to a new challenge outside the project - but still within the realms of Open Source. Kevin has

Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer

Hey all, I have a setup with two analog lines coming into and Asterisk 13 box with a TDM400P and it takes a lot of rings before asterisk takes over. I've traced this same box on two different analog providers so it probably isn't a problem with them. I DO have callerid enabled and not sure I can turn it off (if they will let me). Any other ideas of making it ring through faster? By the

On several occassions lately, my home Asterisk box has stopped registering with my VoIP provider. I haven't been able to reproduce the problem, and the log doesn't contain anything useful. How can I increase the log verbosity for SIP registration-related events? I've looked through logger.conf and tried searching with Google, but I haven't been able to find a clear answer. This

I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). What I found curious was the caller's name was Asterisk, and the caller's number was Asterisk at 10 or or Astrk at 10 similar. (I don't recall the exact

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

Background: I dabbled with asterisk years ago, and more recently have more-or-less functioning IncrediblePBX systems for experimenting, but I want to understand more so I'm now working with distro packages (Ubuntu) and hand edited configurations files. I have three SIP "trunks", each providing me with a UK telephone number. They are for my home (1), my wife's home-based

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

Asterisk Users, Developers, Gurus, Etc., AstriCon Europe 2006 is only 20 days away! Sign up now to ensure a seat at the European Asterisk event of the year! Events will be held in Berlin (June 19 - 20), Paris (June 22 - 23) and London (June 26 - 27). Don't forget to hold the date for AstriCon 2006 USA in Dallas, October 24 - 27. REGISTER: http://www.astricon.net/ TUTORIALS: - Asterisk

Hi folks, Going to astricon this year? Feeling a bit nervous as planning to take the exam this time. Any one else doing the same? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090814/092dda7f/attachment.htm

Hello, Can someone outline what tutorials will be covered at this year's AstiCon in AZ? Are the tutorials going to be worthwhile for fellow Asterisk users/admins that have been actively building, running, and administering Asterisk boxes for years? I have a suggestion for one demo that would certainly interest me and other more advanced users. I would *love* to see a hands on live

Hi list, I know it's a bit OT, but for those who will be at the Astricon, we are organizing a very informal meeting (maybe in front of a pint or two) to talk about Asterisk for call-centers. No marketing or anything - just a way to exchange ideas and meet f2f. I created a facebook group to organize it - see https://www.facebook.com/groups/507826572618269/ See you in Atlanta! l. --

Hi All, Today on the VUC, we'll be welcoming Sipgate CEO Thilo Salmon back to tell about their choice of partners in their latest services. I will be announcing the first #VUC VoIP & Tell discount code for Astricon in Denver, October 25-27. Join us on sip:200909 at login.zipdx.com using g722 or g711 codec or see http://vuc.me for the other ways to connect. Hear you there! /r

AstriCon 2004 is now a reality. We have secured our conference center hotel (the Atlanta Marriott Century Center) and will be opening the website for registration next Monday, June 28th 2004. The event will be held September 22nd, 23rd and 24th and should provide a wealth of information for every Asterisk user. We finally have solid pricing information as well: 3 Day Asterisk Package $400.00

Registration to Astricon - the first Asterisk user's and developer's conference - is now open. Astricon is taking place at the Atlanta Marriot September 22-24. Digium is our Diamond partner in arranging this conference. The web site is updated with information on hotel, prices and speakers for the tutorial day. The agenda for the conference day will be published soon. (Speakers: Mail me

> > Message: 2 > Subject: RE: [Asterisk-Users] Astricon costs... > From: Steven Critchfield <critch@basesys.com> > To: asterisk-users@lists.digium.com > Date: Wed, 21 Jul 2004 21:30:26 -0500 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-07-21 at 19:34, Steven Sokol wrote: > > > Has anyone really looked at the costs for Astricon. But the

-- GOOD NEWS! -- As ground transportation and the costs associated therewith are something of an issue. So, in the spirit of making AstriCon available I've contacted the hotel and arranged/discovered some low cost options: -- MARTA -- MARTA, the Atlanta mass-transit system (i.e. train) can get you there. Take MARTA from the airport north to the Brookhaven station. The hotel shuttle will