similar to: VM low volume - testers needed

Hi all, I use asterisk cvs-head from last week and was wondering why my macros are not working like I expected. A macro will be aborted if one application returns error (-1), but some applications (like ChanIsAvail() in my case) do return -1. So when I want to use ChanIsAvail() in a macro, it does not work if the channel is not available, because the macro is not executed on n+101. The macro

Hello - I have a TDM400P with 2 FXO and 2 FXS modules. Feeding the FXS modules are two VOIP lines which are terminated by VOIP adapters and have regular RJ11 wires connecting to the FXS ports. Since the two different VOIP lines have different phone numbers, and I know and can tell asterisk which VOIP line is connected to which FXS port, can I cause a distinctive ring on the extensions if a call

I hope the subject isn't too buzzword compliant :) I'm just curious: What have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with. Thanks! -Brian

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

Hello everyone. I'm new to Asterisk but got some basic functionality going last night and I'm just giddy to have my own PBX ;-) Sorry if these are silly questions: My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a very basic PSTN line coming in from the phone company, I tried to get the most no-frills line possible (didn't pay for caller ID, voice mail, etc.). I

Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box

Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten => 400,1,Answer() exten => 400,n,Wait,1 exten => 400,n,SayUnixTime(,EST5EDT,) exten => 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious!

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone.

Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be

I looked on the voip-info wiki and found sparse and conflicting information on how to do this with Asterisk... My incoming lines are all on Zaptel. Is there a simple why to implement a '*363" (do not disturb) toggle via the dialplan? It would be nice to be able to pick up an extension, dial *363, and have all calls sent to voicemail without ringing the extensions. Doing it again would

Greetings rsync gurus! I'm trying to run rsync from machine A to backup files on machine B. I want the files to be copied into a directory -- /backups/B -- on machine A and mimic the directory structure on machine B. Machine B has the following file systems (and some NFS mounts I don't want): / /boot /usr /usr/local /usr/local/apache The problem comes when I run rsync with

Hello List, After seeing a few positive responses for the Linksys SPA-942 phones I was hoping to get some answers on the following questions: * How do the phones handling system wide paging? Is it similar to the Polycom phones? * Can a corporate directory be configured with the phones using Asterisk? * How is the speaker phone quality? Thanks Roy Anciso Director of Technology Manistee

OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line

I've just Replaygained several of my Vorbis albums with Vorbisgain, ranging from 80's metal to present day soft rock. What I don't understand is why does Vorbisgain actually make all tracks QUIETER? I see an average of -7db on most albums. And after that, not only are they substantially quieter than my MP3s (which is a pain), but it also fails to really "even out" the volume.

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

Hello, I've read in a thread from this mailing list ( http://lists.xiph.org/pipermail/vorbis-dev/2009-September/020019.html) that the usage of FLOOR1_fromdB_LOOKUP table is to convert Floor1 values from a dB scale to a linear scale. In a dB scale, given a reference value, the other dB values are obtained from it, normally dB_value = 10log(p1/p0), where p0 is the reference value. Supposing p0

Hi. Since I'm new to this mailing list, I don't know whether this was discussed in this mailing list, or if it is already fixed. Anyway, the problem is that all audio encoders out there add a small amount of silence to the start/end of the encoded file. This is usually a small amount, around 0.03 seconds. Most of the time it goes unnoticed, BUT when you have tracks that flow into one

Hello, I'm new to R. In Sapa package there's an example for function SDF like data <- as.numeric(sunspots) methods <- c("direct","wosa","multitaper","lag window") S <- lapply(methods, function(x, data) SDF(data, method=x), data) x <- attr(S[[1]], "frequency")[-1] y <- lapply(S,function(x) decibel(as.vector(x)[-1])) which