search for: faraz

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...

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

...List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <dcc007e10803151320s3498a7ecg9837e5a94a36389 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Fri, Mar 7, 2008 at 9:52 AM, Faraz Khan <faraz.khan at emergen.biz> > wrote: > > > It does work. Did you do the switch statement in extensions.conf? > > > > If not check voip-info for "Asterisk Realtime Extensions" > > > > Hi Faraz, > > I just realised I never replied to t...

...e the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz

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

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

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

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

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

hm.... I think, that is valid in c but next code too doesn't works right: ; ModuleID = 'sample.lz' @.str1 = internal global [6 x i8] c"world\00" ; <[6 x i8]*> [#uses=1] @.str2 = internal global [7 x i8] c"hello \00" ; <[7 x i8]*> [#uses=1] @.str7 = internal global [7 x i8] c"father\00" ; <[7 x i8]*> [#uses=1]

...cently 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 Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz

...central management). I want the phones to authenticate/register to the central server but use the appliance as a 'proxy' so that all calls are sent through it only (and IAX2 trunked, obviously). Is this somehow possible or do I HAVE to register my phones to the local appliance? -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz

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

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

...es this issue. Its the proxy authorization message that really pisses the cisco off. 4. I have tried randomly to use insecure, fromdomain, etc but nothing works. Also- the same phones work perfectly with our Asterisk 1.4.26 office server. Help is most appreciated. We are at a loss here. -- Faraz R Khan CEO Emergen Consulting Pvt Ltd www.emergen.biz +92.21.529.0381 (3 lines) x200

the ONLY thing I cannot do is file/open and file/save - file/open doesnt do jack shit and file/save keeps telling me I have a incorrect filename... anybody else? is this a general problem or Im messing something up?

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

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

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