similar to: SIP URI's: possible now?

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

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

Mark and Ed received word today that the long-awaited RFC for IAX2 has been approved by the IETF, and is now published: http://www.rfc-editor.org/authors/rfc5456.txt Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny Shumard! Lots of revisions and discussions have paid off. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open

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'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 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'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

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

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

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

ok all. I have sent the PHP web gui code to Mark at Digium but have not heard back yet. I dont know that status if he wants to CVS it or not. maybe if he does not answer in a this next week ill just upload it to a website for all to DL. we could add to this as a base and get it to work better for all again the current URL for it is http://rads.netcom.utah.edu/openconf/openconf.php Dave P

Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan

Here's a clip of comments lifted from a Cisco bug list. This will be perhaps useful to those of you who have just purchased a Cisco phone off eBay. JT ------------- (1) Short problem description: Documentation on how to load SIP image on phone with skinny software (2) Longer problem description (what happens): If the phone is loaded with the Cisco Skinny code, then there is a small

I seem to be having a conceptual problem with the "While" and "EndWhile" applications. It seems that on the first cycle, even if the result of the "While" is false that the enclosed applications will get run. Is this expected? It seems to be counter-intuitive, but I don't know what the intent of the While routines is. I could of course put a