similar to: Asterisk System Setup Question

I've been trying to post this message to the list, but it keeps getting bounced back to me. Sorry if duplicate messages are coming through. 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

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

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?

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

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 >

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

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 all, I was just on the phone with a B2C company in Australia who are looking to integrate an Asterisk solution with their Salesforce.com CRM platform. They are looking for a consultant/team to provide the following functionality * Complex IVR Eg can interface via API into Salesforce for customer service interaction and product initiation. * Call centre Eg basic queues,

Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews

I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/88e22671/attachment.htm

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

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 Problem: Clients are unable to authenticate with Active Directory credentials, receiving a "password incorrect" error. Verification: sudo net ads testjoin shows a successful join. wbinfo --ping-dc confirms

Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

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 impossible today. I agree! Also, something I've been looking for is a way to pull

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

Sorry if this dupes, didn't seem to post to the list when I emailed it a few hours ago. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ________________________________ From: Dean Collins Sent: Thursday, December 06, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: "Happy