similar to: Skip Announcement Confirmation in MeetMe

Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg

I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Is is possible to specify the MOH Class when defining a MeetMe extension? I tried exten => 300,1,MeetMe(300|M(class)) But that did not work. Thx, -Rob.

Does anyone know how to suppress the "Missed Calls" indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg

Hi, is there a way to use g729 in meetme? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070929/74f6e5d9/attachment.htm

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

On Wed, 4 May 2016, Nux! wrote: > Direct links > > https://www.imagemagick.org/discourse-server/viewtopic.php?f=4&t=29588#p132726 > https://bugzilla.redhat.com/show_bug.cgi?id=CVE-2016-3714 > > Mitigation: > > As a workaround the /etc/ImageMagick/policy.xml file can be edited to disable > processing of MVG, HTTPS, EPHEMERAL and MSL commands within image files,

hi everybody, we have problems using wincvs with a cvsroot on a samba-server. we think that the problem is the renaming of files where the cvs user does not have write-permisson on: this is our configuration: fileserver on a irix6.5 running samba 1.9.18p8 having a share on / named 'coder'. win95 box, running wincvs, setting cvsroot something like

I have the following in extensions.conf: exten => 1000,1,Meetme(|dMic|) According to the 'show application meetme' docs: 'i' - announce user join/leave (new in Asterisk 1.2) Well, when users join the conference, Asterisk records their name, but does not broadcast it into the conference. I have Asterisk version 1.2.4. I know this has worked in the past. This sure as heck

I googled on this for about an hour and the most relevant hit I got was, of course, the first hit: http://www.sowerbutts.com/linux-mac-mini/#support In it, he indicates that the stock Bluetooth module "should work, but untested" - he doesn't qualify the statement with anything. Has anyone tried chan_bluetooth or even the Bluz stack on a Mini or a G5? If so, under Linux or OSX?

I am trying to reshape a data set. Could someone please help me with the reshape, cast, and melt functions? I am new to R and I have tried reading up on how to use the reshape package, but I am very confused. Here is an example of what I am trying to do: subject coder score time [1,] 1 1 20 5 [2,] 1 2 30 4 [3,] 2 3 10 10 [4,] 2 2

Hi, I've trying to decode a FLAC audio stream. I have a reader which sends raw byte data to my FLAC wrapper class. Only once the decode function below returns, the reader will send new data. Hence I want to decode until the stream is empty, but I will add new data to the stream once it is empty. *void **MyFlacCoder::**decode(char *data, int bytes) { mBuffer = input;

Hello, I have a rather annoying issue on going with one of my centos virtual servers. the server hosts a website using apache and mysql ,there are three persons involved with keeping the site up and running. and i am his root due to the fact he does not know anything with about Linux. there is an php/sql coder , and the site owner which only knows to use the CMS and upload new articles to the

Hello all, As I'm sure is pretty common, I have some extensions that dial mobile numbers after a local timeout. I would like to prompt the caller to record their name after the local timeout and have the recipient be able to hear the name prior to accepting the call. Recording the message is easy enough, so I thought about doing something like dumping them into MeetMe after they record

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

I've come into some code that produces different results under R 2.1.1 and R 2.2.1. I'm really unfamiliar with the libraries in question (MASS and nlme), so I don't know if this is a bug in my code, or a regression in R. If it's a bug on my end, I'd appreciate any advice on potential causes and relevant documentation. The code: