similar to: Asterisk System Setup Question - 2nd Try

Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones

Hi all, I'm not getting any sidetone on my Snom 370. I searched the web and the snom wiki, but I don't see any place to enable/adjust it. Callers say I sound great on the other end, but I don't hear myself so it is a little off-putting. Any suggestions would be appreciated. On a related note, some times (maybe 1 out of 10 calls) I get the side tone, but its delayed by a second or

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

Hi list, Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. Thanks... Raul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071218/c543b407/attachment.htm

Thank you. While I'm currently in the process of migrating to AlmaLinux 8 via an in-place upgrade, I initially wanted to update the base packages on CentOS 7 before proceeding. However, given the circumstances, it seems deploying a separate AlmaLinux machine for testing might be the most efficient approach to troubleshooting the issue. If the problem persists on a fresh AlmaLinux

On Fri, 2001-11-16 at 19:29, Samuel Hathaway wrote: > On Fri, 16 Nov 2001, Jerome Alet wrote: > > > one thing that would be nice in DarkIce would be to allow the user to pass > > specific reencoding options for each server, e.g. DarkIce could acquire > > the audio in stereo and send it to a server in mono and in stereo to > > another server, which is AFAICT

We've got our Polycom phones auto-answering for paging. Is it possible to configure a PAP2 to auto-answer for either paging or intercom? If so, how?

OK this is the RTFM question of the day but I need a sanity check. I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection. 2 Aastra 51i---------| |-NAT on dsl moden--(Internet)--Asterisk PAP2t----------------| The DSL modem/router which has QOS set for the src and dest to the * box the PAP2 has both lines registered @ ports 5060 and 5061 and work like a charm. one of the

Hey I want to generate a discrete uniform distribution as follows: For example: I want to get 278734 records each with a numbers between 7-10. And the sum of numbers in 278734 records to be equal to 2253712. Once this is done, I want to get that printed to an excel file such that Record Value 1 9 2 7 so on 278734 8 The

Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan ######### [root at localhost ~]# yum update asterisk18* -x asterisknow-version Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * base:

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

On 16 Nov 2001, Zaheer Merali wrote: > That is an idea that is coming up in the next ZStreamCaster. > ZStreamcaster 0.1 currently allows you to save a stream to disk at a > higher bitrate than you send to the icecast server at. > > I am planning to add a feature that allows you to have n streams going > out, each for different bitrates (or alternatively different sample >

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as

Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisknow at lists.digium.com Hello. Can someone tell me why AsteriskNow is reverting to registering

Hi All, I'm having a bit of trouble with my new *NOW setup. I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from SimpleSignal.com. Outbound calling works great, but I'm having some trouble with inbound calls. First, we would get the "the number you have dialed is not in service" error on inbound calls. After some googling, I found out that I needed

Guys, please help me in understanding what I'm mistaking... Description: I've configured my AsteriskNOW (beta 6) server, in service providers section, with the parameters provided by my ITSP. Until now I've used this configuration with SJphone and all worked perfectly. Now I've decided to use this account with AsteriskNOW to begin my experience with a VoIP based PBX. The

On Thu, 4 Apr 2024 14:28:16 +0100 Zaheer Abbas via samba <samba at lists.samba.org> wrote: > Hello everyone, > > Samba stopped authenticating AD users after minor upgrade. > > Environment: > - OS: CentOS 7 > - Samba Version: Upgraded from samba-4.10.16-15 to samba-4.10.16-25 > The problem is, Centos 7 will go EOL in about 3 months, at which point you will have to

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message "login incorrect". I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up