similar to: Incoming Call trouble with new *Now 1.5 setup

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 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 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

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

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

With some carriers the initial Audio (2-4 secs) seems to get cut off when using a Auto Attendant or Conf Meetme. Is there any known remedies for that. Just want to know if others have seen that esp. with Level 3. If Auto Attendant says - "Welcome to ABC bank" Caller only hears "Bank" -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 >

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

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?

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

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

>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 >rates and channels). While we're all

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

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

Hi all, Sorry for the VP8 question on the Theora list, but I think all the relevant people are here :) I'm using Icecast to distribute some Ogg Theora streams at the moment with a view to adding VP8 along side these in future. Ideally I'd just use Ogg VP8+Vorbis, I know there is a mapping for this already but how much support from the browsers can I expect for this configuration?

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. I have seen mentions of Skype integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for

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 everyone, after a long wait, Cortado 0.2.0 has finally been released. This release adds support for seeking in static files, and numerous bug fixes. Full release notes attached. Enjoy ! Thomas -------------- next part -------------- Release notes for Cortado?0.2.0 "Broken Record" Cortado applet Features of this release * Seeking in on-demand files and