similar to: playback in [macro-stdexten] problem

I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware version (Application Code Version: DTA version 1.0 US (8x8 001111)) onto it via TFTP, so I could access the SIP configuration. Under the SIP config, I put the IP of my * system, the 5060 port, and for Domain Name, I put default (is that right?). I checked off the Send Registration Request box. Dial Plan I left at the default,

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 am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

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

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

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 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

Hi! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

I'm looking for a function in R that extends kappa to multiple raters when there is more than one response per subject. For example, say a group of doctors have to assign diseases to patients. Each patient will be assigned one to many diseases, and the number of doctors assigning diseases to any one patient will be two to many. Here's an extremely simple example of the type of data I

Consider the following data frame X <- data.frame(Titanic) Does anyone know of an easy way to convert X into a multidimensional array? Example that doesn't work X <- as.array(X, dim=c(4,2,2,2)) To do what I need, X needs to be converted into an array of dimensions c(4,2,2,2) in this case, not a table. Thanks in advance.

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

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

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

Hi, I have a data file with columns separated by ";" I read this file without any problem using read.csv2( ) but I had problems trying to read it with read.table( ... sep=";"). So it is not a problem for me, but I wonder if there is a bug here. drt <- read.csv2("t.txt", header=TRUE) # ok dcs <- read.table("t.txt", header=TRUE,

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to