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...rom: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23 June 2004 10:33 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb john lawler [asterisk-users.lists.digium.com@tgice.com] wrote: > You don't have to put this in the dialplan. It's one of > those low-level functions in Asterisk (possibly controlled at > the driver level-- I'm not sure about that). If you have an > extension defined, pick up the handset and dial '*78', you > sho...

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

> -----Original Message----- > From: john lawler [mailto:maillist@tgice.com] > Sent: Wednesday, January 07, 2004 1:38 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] yet another question on DID trunks > > Hey Steven, > > Sorry to bother you yet again w/ a question on my seemingly endless > quest to get DID trunks setup for...

Is there any way to take an incoming callerid string and remove the given "name" part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Thanks, John Lawler

Hi guys, This is a two part question about the Voicemail application: Firstly, is there a way built in to the current app which would allow me to have Asterisk call a phone number everytime a certain mailbox receives a new voicemail? I know about the email and pager notifications that are already built in to voicemail.conf, and I've used those but also have a need to do what I describe

I'm having major problems routing a modem (data) call through my Asterisk box. I've got a single incoming POTS line through my X100P, and a few extensions (and modems) plugged into the ports on my TDM400P. I've been using the system for a few weeks for voice applications and everything is working pretty well in that area. But just recently I was trying to place a data call on

Hi guys, I've been running 0.5.0, which is dated sometime in September of this year and I've noticed a couple of new features in more recent code that I'd like to use, but am hesitant to go w/ CVS code. My system is not exactly a production system, it's mostly test, but I'm still leery of the fresh code. I'm wondering when the next stable release might come out, and

Hi guys, I think I posted on this issue before, but didn't get a response. I've still not been able to resolve the issue. I've got a small installation of Asterisk running one 4 port FXS Digium card and 1 FXO Digium card. I'm having difficulty routing modem call through one of the extensions out through the FXO card. By difficulty, I mean it won't work. The calls

Hi guys, For the first time, I'm attempting to run asterisk as a non-root user for all of the obvious reasons. I'm attempting this with asterisk-1.0-RC2, based on the fairly straightforward directions found here: http://voip-info.org/tiki-index.php?page=Asterisk+non-root The only problem I can't get figured out is my mpg123 processes not being spawned properly. There's

Hi guys, I've got a quick feature suggestion to solve a problem that I don't think is readily solvable having to do with an "after hours" message, playing only during (or rather outside) specified times. I know all about the helpful feature that already exists which allows you to tack on a cron-ish specification of time, date, month, etc. to the end of an "include

It looks like this issue has been raised before, but I see it mostly ignored and no answers given, so here it is again: For the past couple of years, I've wanted a queue that works very similarly to roundrobin/rrmemory, but that doesn't remember anything about where the last ring went to. This new strategy would always start at the first member (as defined in queues.conf) when a new

Hi, I'm still pulling my hair out getting my queues setup in 1.2.13. I went in to implement my custom "roundrobinreset" strategy (mentioned in a post by me here: http://lists.digium.com/pipermail/asterisk-users/2006-October/170713.html and a similar issue is addressed by the developers back in May: http://lists.digium.com/pipermail/asterisk-dev/2006-May/020916.html and I got

Hi guys, I just moved from Asterisk release 0.5.0 to CVS 2003-12-22, and after overcoming a few changes in my configuration, I encountered one problem that I couldn't shake that was working fine in 0.5.0. It's the fax detection. I just have a simple extension setup like this: exten => fax,1,Dial(Zap/4,30,tr) exten => fax,2,Hangup in my main incoming context. This used to

I've got a couple of different situations where I'd like to do something like zapbarge into a specific channel but I'd like to be able to actually talk to the party or parties on the channels, not just listen like w/ zapbarge. There are two scenarios I can think of right now where it'd be very handy. a) when the outside line is ringing and Asterisk is waiting to answer or

I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not having much success yet. First, I noticed that nothing happened even after I had enabled all of the options in zapata.conf & setup a sample extension in extensions.conf. Then I read something about how Asterisk uses mpg123 to play the files. I discovered that this had not been installed on my system, so I

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's<extension>' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup, even if the handset is hungup, the channel remains frozen

Hi guys, I've had a sporadic problem recently with one of my users on our POTS line. About 1/3 of the time he dials a number (usually from a speeddial on his phone, I think), he'll get some phone company message (from the outside) about how the call could not be completed as dialed or something like that. However, the logs (and the console) always show the correct dialed digits.

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.