similar to: CVS and Releases

Thanks to everyone for your quick responses to this question. I'm very excited about the Asterisk project, and the growing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From:

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

I realize this is a shot in the dark, but I'm trying to find a VoIP provider that offers 402 or 712 area code DID numbers. I'm almost completely convinced that no one offers these area codes (eastern Nebraska, western Iowa), however considering the wide audience of this mailing list I thought this would be a good place to ask. I would prefer a provider that allows for Asterisk use, but I

Dear Asterisk Gurus: Our county is finally ready to begin implementing IP telephony. We intend to use a Cisco router as our PSTN gateway and Asterisk as our soft switch. The plan is to use SIP between the Cisco router and Asterisk. We will have a single PRI T1 connected to the Cisco router for PSTN access. My question is this: Are Cisco routers able to pass caller ID information (from PRI

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten =>

> From: Philipp von Klitzing > Sent: Tuesday, 02 December, 2003 10:50 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] How to restart * thru phone "when convenient" > > You could use "at" to issue the command at a deferred time. > Yes, sure, but this ain't that nice "asterisk only". :-> You should be able to place

> -----Original Message----- > From: Jared Smith [mailto:jsmith@drgutah.com] > Sent: Monday, 12 January, 2004 10:41 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Turning a profit (WAS: More words > for Allison) > > > On Mon, 2004-01-12 at 04:49, Alastair Maw wrote: > > Hmmm... I think John's turning a profit... :) > > That was my

> I have placed "ignorepat => 9" in just about every context I > can think of in my extensions.conf, but yet when I dial 9 > from my sip devices, the dialtone is broken. I even tried a > nearly untouched version of samples, and it stil doesn't > work. Is there something somewhere else that needs to be set > to make this work properly, like may in the sip

For anybody who didn't know there is an article on asterisk in February's Linux Journal. AJ

I saw it also on some of my Ubuntu builds, but the issue went away after a make clean/make, so maybe give that a try. Best, luke On Wed, 24 Apr 2024, Ben Bolker wrote: > I'm using bleeding-edge R-devel, so maybe my build is weird. Can anyone > else reproduce this? > > View() seems to crash on just about anything. > > View(1:3) > *** stack smashing detected ***:

Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of

As things stand now, package tarballs with vignettes that are built with R-devel will not install in R 3.4.x, so CRAN can't accept them and someone running R CMD check --as-cran should be told that. A WARNING is appropriate. Most likely what will change soon is that build/version.rds will be saved with serialization version = 2 and this warning will not be triggered just by having a vignette.

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

Dear Rers, is there a way to color counties on a full US map based on a criterion one establishes (i.e., all counties I assign the same number should be the same color)? I explored a bit and looks like the package "maps" might be of help. library(maps) One could get a map of the US: map('usa') One could get countries within a US state: map('county', 'iowa', fill

Hi, I just (~1100h CET) pulled the current wine release from CVS (as I do almost every morning ;-), compiled and now my Lotus Notes client does not complete its startup anymore (yes, I know CVS is bleeding edge...) I get lots of fixme:file:LockFile not implemented in server fixme:file:UnlockFile not implemented in server messages, the window is being created, the menu bar is being drawn and

What cvs versions are people using in a production environment? Is there a good way to choose a cvs version, like maybe one right before a major change? Are there any plans on having a "testing" branch that is not as stable as stable, but not as bleeding edge as head? I'm thinking of compiling a list of specific cvs versions that are being successfully deployed, would this be

Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later..

It's the day before Christmas here in Sweden, actually the night before at this time... We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into "merry-christmas-mode" with the yet undocumented CLI command "frosty-mode on", a mode where the PBX will connect all incoming extensions to the "ho-ho-ho" sound file and then randomly pick a

Why not put all this into a pre-commit filter in CVS and be done with it? We'd never be bothered with it again as it would never be committed again. Reid. On Thu, 2005-04-21 at 15:11 -0500, Misha Brukman wrote: > Dear LLVMers, > > If you live on the bleeding edge (i.e. CVS version), please read! > > On Wed, Apr 20, 2005 at 12:12:54PM +0200, Markus F.X.J. Oberhumer wrote: