similar to: Configure WAN Phone

I am trying to configure mysql to use a different datadir than default in order to move this to a larger volume. I have copied all mysql data from /var/lib/mysql to my new volume and ran both chown -R mysql:mysql * and chmod -R 660 * in order to setup correct ownership and rights for the data. It was working for a few days until today upon going into mysql and typing show databases, I receive

Same here. I called them yesterday plus email and still no reply. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett-asterisk@worldcall.net Sent: Friday, January 21, 2005 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and

Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan

On 30 April 2017 at 16:54, Tech Support <asterisk at voipbusiness.us> wrote: > I thought this was a non-commercial list. > > Yeah, I wouldn't mind so much if it had actually answered the original poster's query. "Switch to our proprietary solution and we can offer you this proprietary solution" isn't a contribution, it's an ad. -Barry > >

Thirdlane Connect can be used as a softphone. It works in modern browsers (no installation is required), on Mac, Windows and Linux desktops, and on mobile phones. Besides basic softphone functionality, it provides instant messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk,

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming

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 Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am having a big issue that i just cant get fixed for a while now and its causing me lots of grief. It seems like since we got FIOS installed (including switching to fios phone lines which are supposed to be the same on our end) i am having massive problems with asterisk not hanging up dead calls for days, even weeks if i dont catch it. It slowly builds up randomly not ending a call

I am using Verizon FIOS to my home. I subscribe to a 5 MB down 2 MB up data package. I continue to pay for a standard voice line in addition to the broadband connection only for local calling, fax and emergency 911 use. The way it works is that the fiber optic connection is terminated on the house via an ONT (optical network termination). The ONT can provide connectivity for three types of

Greetings, A long time ago I setup a Linux machine as a Gateway/LAN Server using Verizon DSL as the ISP. I used the following HOWTO as the guide - DSL HOWTO For Linux: http://www.tldp.org/HOWTO/DSL-HOWTO/index.html Is there something comprable for Verizon FIOS? My Gateway machine runs Fedora. For a new server, I'm considering setting up a CentOS machine, while still using Fedora on my

http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for "eye candy configuration" of Asterisk. However it costs lost of $, and I'm currently only an "experimenter" so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium

http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for "eye candy configuration" of Asterisk. However it costs lost of $, and I'm currently only an "experimenter" so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium

You can use "Command" command, and "sip show subscriptions" as a parameter -- Alex Epshteyn email: alex at thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 ----- Original Message ----- > From: "Leandro Dardini" <ldardini at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at

Hello: Ed Zaleski ran some remarkable and very impressive load tests on icecast reported on www.icecast.org (Home page). What wasn't mentioned in the test specifications was the type of internet line that was used which totally influences bandwidth and subesequent overall load characteristics. Questions: 1. Was it a T1 line, Verizon FIOS, or cable internet, or what? 2. What were the

I have two Asterisk Systems. One on of those, when I execute this: Dial("SIP/teleglobe-007931d0", "SIP/13033372500 at teleglobe|60|oL(4007520000:60000:30000)") ... It causes Asterisk to immediately read out the time limit of the call (66,792 minutes), as soon as the other end answers, even though we aren't down to 60s remaining yet. Asterisk then goes into an infinite