similar to: Large-scale voicemail deployment: any experiences?

I ran across this URL the other day: http://www.advancedvoip.com/ It seems that they have a low-priced solution for billing, which seems to have some interesting features (at least, via the canned demo they provide.) It handles CDRs and also (sorry, Jeremy) Radius, and seems to have a lot of features. I'll be calling them later today to discuss their product, if only out of curiosity

I'd like to float the idea of a VON panel discussion for large-scale open-source deployments, specifically using Asterisk as an application service and as a gateway service. In order to do that, I'd need to get a list of panelists together who might be interested in speaking. If you manage an installed base of >1000 users, and you're using Asterisk, I'd be interested in

Hello! I've been asked to get a show of hands for some analysts for users in Higher Ed - Universities, Colleges, or any other 2 or 4 year degree- granting institutions. If this fits you, please let me know your contact data and briefly how you're using Asterisk, and if you don't mind I can pass your contact data along (for consumption by humans only - this is not a mailing

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the "common" features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

Hello all - Second-to-last notice to collect RSVP's for interested parties for the Asterisk Birds of a Feather dinner/beer meeting. There are five people signed up for this at the moment. Looks like it's going to be the 23rd of September, since that fits well with the VON conference schedule. Please email me directly for details. JT >Date: Thu, 21 Aug 2003 01:52:52 -0700

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

I'll post my semi-annual request for press contacts. If you have an Asterisk installation that matches one or more of these adjectives: - enterprise-oriented - government-oriented - education-oriented - unique - clever - large - complex ...we would be interested in having you talk with the press! We at Digium keep a list of people who wouldn't mind talking to the

Hi Everyone ! I have been using Asterisk for 2 to 3 months now. I have made peer to peer calls using Asterisk both with H.323 and SIP. Now there is a requirement that I have to setup asterisk in my office supporting upto a 1000 extensions (pstn or IP endpoints). Now the question that I wanted to ask was that is it possible to deploy Asterisk with so many extensions ? Has anyone ever used

I'm wondering if there is any command-line interface available for working with values stored in astdb. Of course, I can run "asterisk -rx "database show" " or other commands like that, but I was hoping for a local command that would allow manipulation or output in some other form. Is astdb in a standard db format? JT

This is a fairly large list of hardware phones, features, and URLs, with some vendors I've never heard of before on it. I suspect that any phone that doesn't explicitly say "SIP" is an H.323 phone. http://www.tmcnet.com/it/0403/0403sg.htm#sidebar2 JT

I have several phones running Cisco's 6.0 SIP software release at this time. Two of the phones have not shown any abnormal behaviors, but one of them has an unsettling propensity to lock up after several hours, where the softkey labels disappear and the phone stops registering, requiring the standard *-6-settings reboot sequence. Otherwise, the phone seems to work OK except for a slight

This is not specifically Asterisk-related, but I think that sometimes a "pre-emptive" clue is a good thing. Viking Electronics (http://www.vikingelectronics.com/) has some neat widgets that attach to phone lines, which I'm sure many of the people on this list would find at least somewhat useful in conjunction with their Asterisk systems. Many of the widgets can be replaced with

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

I've been involved with getting better data for running Asterisk on the Amazon EC2 cloud computing system. Here are some calculations I've made on costs based on current published prices on Amazon's system. Feel free to tell me that I'm wrong with these calculations - but be specific if you find any problems, as I suspect others may glom onto these figures as gospel and

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

If the 1/2 ATA186 device supported IAX I think I'd have found nirvana :-)(especially at $49.95 ;-) Lenny -----Original Message----- From: John Todd [mailto:jtodd@loligo.com] Sent: Monday, April 07, 2003 7:50 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] IP hard phone devices (grandstream) The device at VON would have various parts of it's LCD screen go into