similar to: Advice on best operator phone (with attendant console)

Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati ons/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 roy at

If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing

Dear all, Just wanted to know if any one had deployed the Grandstream GXW 4024 yet. Wanted to hear any feedback and/or problems with this unit that you may have experienced. Thank you. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey

Dear All, We have a customer who is opening a new office in Dubai and we know that VoIP is blocked over there. Has anyone a solution to getting VoIP back out (we want interoffice calls back to the UK)? We we're thinking of IAX trunking, but not sure if that is blocked or just SIP etc. A VPN works, but is not great. We have seen: http://www.speed-voip.com/voiceguard.html At the moment it

somebody knows some application web that allows me to call to my internal extensions of my asterisk, example click to call. I was proving the click to call of this example but it doesn't work http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html greeting rickygm

Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm

Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you.

Hi, I had asterisk 1.4.17 with the extensions which is configured in the sip.conf it was working fine. Now I am having the requirement to authenticate the SIP users through the OpenLDAP not through the sip.conf. Steps I have done : Did a check out by using the following command, http://svn.digium.com/svn/asterisk/trunk. [^] then given configure, make , make install. and taken the sample ldap

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

Hello, How can I make a live chat (mainly text, but with voice/video chat if possible) interacting with asterisk? Can asterisk control simultaneously the queue between people calling by phone and people by web chat? At my work, there is a call center using asterisk to control the queue of the clients (by phone) already. This part is ok. But now I need to make a chat room at the website

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

Guys, I assure you this is probably the most interesting and weird problem you have encountered (or definitely up there). I'm using ABE 2.1.2C and roughly 500 or so Cisco 7911G Phones. The following is what happens: When trying to dial a number from the cisco 7911G phone it may randomly get stuck on 'Dialing'. The SIP history on the asterisk end goes like this: 1. Cisco ->

Thought i would share this so it doesnt annoy others as much as it did me :) If you recently installed a GXW 40XX and your extensions start ringing magically now (ringing for no reason, pick it up its a clear tone) you need to check the "Disable send MWI" in your gateway. apparently certain old phones do not like the MWI signal and treat it like a ring tone. -- Faraz R Khan

Dear all, Wanted some help on a solution we wish to deploy. There is a central Asterisk server which is connected by some 30+ remote sites. Each site has a substantial number of users within them (200-300). What I wish to do is save on bandwidth by trunking connections between those sites and the main site. I can use a flash based solid state device for this purpose (Xorcom or ASterisk

Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 "lines" registered w/ the server? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com ________________________________ This e-mail, facsimile, or letter and any files or

JR Richardson Engineering for the Masses> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of asterisk-users- > request at lists.digium.com > Sent: Saturday, March 01, 2008 12:00 PM > To: asterisk-users at lists.digium.com > Subject: asterisk-users Digest, Vol 44, Issue 1 > >