similar to: Dialplan Sanity Check please

hey guys I thought I was making progress on my dialplan when I realized that the class features that are available for zap channels aren't available for SIP channels. I see references in the archives to adding pattern matches in the dialplan for CLASS features which has raised a couple questions. 1. Is implementing CLASS like features via the dialplan the currently recommended way to do

How are we all documenting complex dialplan? Is there something similar to Doxygen? I've got around 20 config files covering around 60 contexts and 40 variables. Of course, I've maintained a basic list of the major stuff, and documented the code throughout, but it's grown to the stage where it needs to be better documented, have a proper flowchart etc. Talking of flowcharts, I see

Hi, thanks - that looks really good! I was about to embark on some non-visual stuff using Ragic, but this looks great. Is there a binary anywhere, or any instructions to compile? I've never compiled C# code before, and although a quick google suggests it shouldn't be too hard, I might need to know a few things like what version of .net it should be compiled with. The readme just points

ok, I've got yet another newbie question. My extensions.conf is getting rather longish and I'm getting dizzy moving back and forth editing this thing. Can I use the include command to include a file in order to break extensions.conf up into more manageable pieces ? Is breaking up the extension.conf file an OK thing to do ? Maybe something like this: include

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Does anyone know if Allison has recorded anything along the lines of: "You don't have permission to dial that number." Thanks. --Lance Arbuckle

In the 'me too' vein, I also have an untested 3624 here on the shelf and am waiting on a shipment of T100 cards to play with. Documentation is very hard to come by. Alcatel are certainly the owners of the Mainstreet product line but, without a support contract, any documentation they may have is essentially unavailable as their per-incident fees for support cost more than most of the

Hi. I'm new to Asterisk and have been working on setting up a development server but have gotten myself a bit confused. I'd like to implement the following logic for calls coming from the PSTN: Check for caller-id yes => keep going no => play SIT and prompt for telephone number Check time of day to see if it's day / night day => ring some phones

I had just about about sold a new asterisk phone system to a local company when they called back asking if I could match a proposal from Novergence.com. I haven't seen anything on paper but was told their proposal was to provide a new phone system that would replace the existing 8 line 12 extension system, provide an internet T-1, unlimited local and long distance, voice mail, and two

Ok, first off, Asterisk is the coolest piece of software I have EVER had the pleasure of using in my 8 years of running linux !! and I know I haven't even scratched the surface feature wise. Before I get too excited, I wanted to get all you experts to look at the how I implemented my after hours test. The goal is to prevent the phone from ringing afer certain hours, just go to VM.

I have a large dataset and I would like to make some kind of flowchart from this dataset. The idea is to show rowcounts from data subsets: data1 rows= 100000 | / \ / \ males females, rows=500000 rows=500000 /\ / \

Hello dear R help group, I wish to create a flowchart of my R code (specifically for profiling the connection of different functions). Maybe something similar to this: http://www.fatesoft.com/s2f/ But for R, and that it will be free (open source would be best). After some searching, I found this package: http://cran.r-project.org/web/packages/diagram/ But couldn't find a way to this

You may want to look into RAnalyticFlow: http://www.ef-prime.com/products/ranalyticflow_en/index.html Regards, Frank ******************************************************** Dr.med. F. Bloos, Ph.D./U.W.O. Oberarzt Klinik f. Anästhesiologie und Intensivtherapie Klinikum der Friedrich-Schiller-Universität Jena Erlanger Allee 101 07747 Jena GERMANY Tel: +49 - 3641-932 32 83; Fax: +49 -

I have the standard parking.conf but extension 700 doesn't show up in my dialplan.... Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise.... Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos => 701-705

Thanks to everyone that responded to my channel bank question. Ive decided that the Adit 600 would be a good choice. Then I got to thinking about SIP phones and wondered if their quality has progressed to the point that they can be deployed to customers who "just want their phones to work" and wouldn't tolerate any SIP hickups. As for pricing, I would think the SIP phones would

After reading the recent thread [http://www.xiph.org/archives/vorbis/200211/0057.html] on the vorbis list about bitrate peeling, I was wondering if it would be possible for xiph.org to post a specification, outline, flowchart, or a drawing on a napkin of what the basic operations of bit peeling would entail. If there was some sort of starting point, then it may be a little more inviting for

In a nutshell and very first of all, you need at least 2 internet connections. Those ISP must be willing to setup BGP peering between your routers and theirs. Once that agreement has been made, you need to get their AS Numbers and submit the ASN request located on ARIN's website (http://www.arin.net). After some paperwork and money exchanges, ARIN assigns you an AS number. At that point,

I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for "NAT-ed" clients on different asterisk

On Sun, January 11, 2015 5:16 pm, Keith Keller wrote: > On 2015-01-11, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: >> >> Indeed. Or another system altogether (sihg). I'm just extending your >> thought half a step farther ;-) > > Or going even farther, if you like CentOS but not systemd, do the work > to get CentOS working without it. Unhappy Debian

Dear Lance, It is not work. Any idea? Regards, Eric -----Original Message----- From: Lance Fredrickson [mailto:lancethepants at gmail.com] Sent: Wednesday, February 17, 2016 12:53 AM To: tinc at tinc-vpn.org Subject: Re: Tinc Router Mode - PING RESULT is destination host unreachable On 2/16/2016 9:04 AM, Eric Yau wrote: > Hi Lars, > > Once I modify the firewall FORWARD rule to