similar to: Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.

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

NEWBIE alert: i'm a linux person, not an asterisk person so i'm certainly capable of handling any linux-flavoured solution you can suggest. here's a note i got from a local company i know (some proper names removed): ===== start ===== Now and again our ISP goes down and when it does give us a hicup, the Asterisk system shuts down (not very forgiving). When it shuts down our phone

Hello list, Slightly off the list topic, but I hope I'll get some help here. Somebody wants me to implement for his project a Cisco based VoIP system. I told him that I specialize in Asterisk based systems, but he is not even aware of Asterisk. The requirement of project is such that chances are slim that this firm will consider Asterisk based system. So I told him that though not experienced

Looks like it's time to fork...... > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Lenz > Sent: Monday, August 13, 2007 7:28 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] FW: The trixbox Revolution Continues! Sign >

I am wondering if there is any such thermostat available which can be controlled from Asterisk. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

Greetings, I have created simple conferencing solution before using meetme application, but this times its a little tricky. My client needs a functionality to call multiple extensions to join a conference room. Extensions will ring like in a ring group, and on pick up, user will be either automatically added to the conference room, or maybe I'll program them to enter 9 to accept and 8 to

Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hi Guys, i have a small issue but bothering me, after restarting asterisk (version 1.4 running on centos) i have the following message that comes repeatedly when i am connected to the CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected does any one know how to stop this or if it's a sign of a

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

Hi all, For a top quality setup, I will need to install high quality VoIP switches with QoS and PoE. My potential customer should not have any problem with call quality. Experienced folks, Please advice me what switches to install and at what price. I may need it for upto 100 phones. What else should I consider so that phones work without problem along with the computers on the same network?

Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas,

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice