similar to: Patch testers needed

If you run CVS HEAD, and are using ODBC storage for voicemail, please apply and test the patch from bug 3024 in Mantis. This patch should _not_ change the behavior of your system at all, and that's what needs to be tested. If you can try it, please report the results in the bugnotes. Thanks!

All, As a bug marshal, I have noticed quite a few bugs that seem to get overlooked/ignored due to the fact that they do not have the appropriate information in the bug information fields, or bugnotes. I am, therefore, Reposting an old post from July 26th, 2003, to help clarify the use of the Bugtracker. Sorry if you have read it before; this is just an attempt to re-familiarize

Dear Skinny/SCCP lovers :-) I've just completed & uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... --jan (*1) http://chan-sscp.sf.net (*2) yes, bugtracker is down at the moment, will fix

Wow, you guys are fast. My apologies, twisted. I realize there must have been a reason why it wasn't marked "resolved" and included in the CVS HEAD, but I was under the impression that those who wanted to and have the knowhow could download and apply the patch. Didn't mean to imply you or anyone else *stated* that it was finished, it just seemed from the dialog in the bug report

It's very possible that the Polycom IP600 will work with this. As it is just an implementation of a SIP standard for subscribing to the state of other extensions. As for the feature improvements you might see them from me, but not very likely. It is easier for me to train my customers to hit *8 (I will probably just program a pickup button for them) than it is for me to figure out what I

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

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

In response to the SIP and NAT discussion, I have updated the ticket on the subject that seemed to be getting the most attention: #104. There are enough clueful people here that perhaps someone can come up with a patch that handles NAT in the elegant way that I describe in the bugnotes, as I am but a mere integrator who has limited C skills. In the absence of such a patch being offered, we

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

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:

A company where I work is building an enterprise-grade infrastructure system which enhances the usefulness of VoIP systems on the public Internet. We're looking for a few enterprises which are running Asterisk who would be interested in being Beta customers for our pre-rollout testing. In exchange for being a Beta customer, you would be entitled to free service through December 2005.

So, Tilghman has put a particularly useful patch in the bugtracker: streaming music-on-hold is now supported. You can now specify .mp3 streams to be played back as MOH in the various places where MOH is used. Hopefully, Mark will install into the main CVS tree shortly. http://bugs.digium.com/bug_view_page.php?bug_id=0000413 This allows you to use the very sophisticated mp3 streaming audio

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 doing some quick research on how to get our videos from AstriCon available in a "reasonable" format that allows easy viewing, reduces our bandwidth costs, and allows good tracking for who/where/what is viewing the videos. YouTube seems to have a very nice set of tools and statistics collection methods, and might be perfect EXCEPT.... Their main limitation right now

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