similar to: Asterisk VoIP in Dubai/UAE?

Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060) iend(0x8fd5060) and running on Asterisk 1.4.21.2: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0

Hi, All I am looking for Suppliers/resellers from Dubai - UAE to buy some VoIP products and Digium's TDM cards. could some one send me some contact information in this regards?. mainly I want to buy Hardware SIP phones, VoIP gateways and DTM cards (FXS). Thank you Kumara

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

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...

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

Hi, We have upgraded the openssh 4.6P1 on Solaris 8 servers. After upgrade we get the below error message whenever we execute the remote commands using ssh. Please let me know what the fix is for this. Apr 8 03:03:43 dvsrv10 sshd[25379]: [ID 800047 auth.info] Accepted publickey for osteam from 10.0.93.31 port 35856 ssh2 Apr 8 03:03:50 dvsrv10 sshd[25381]: [ID 800047 auth.error] error:

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets

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

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

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

Hi, How can I make calls from Asterisk client to NEC NEAX 2400 traditional phone ? Is it possible to have a connection between Asterisk and NEC NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP. Please help me to find a solution ;;; Thanks & Regards Ram Kumar Customer Support Engineer Barcode Gulf LLC Dubai , UAE Mobile : + 971 50 5594178 Email : Ramkumar@barcodegulf.net

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

Hi! I'm trying to set up an asterisk based PBX with a TDM400P +2 FXS +2 FXO modules in UAE/Dubai for home switching / voicemailing. I am using the card Asterisk/Zaptel 1.4.0. I want to include a special route when a certain caller calls into via PSTN. The problem is that I cannot detect the Caller ID. I tryed various setting (cidsignalling, cidstart) in my zapata.conf, here is the last

Hello All We need to connect our client's offices located in Dubai and Pakistan. Suggest us some economical solution. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN: kashif__naeem at hotmail.com Gmail: meet.kashif at gmail.com Skype: kashif.naeem 302 Y Commercial Area,

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

Greetings! Anyone here who can refer me to someone who is looking for a system or network engineer for a company based in Dubai, UAE? I have a pretty good experience in networking R&D and ISP environment and have familiarized my self with configuration of various network services (dns, web, sql, ldap, email), network monitoring tools, mrtg, cacti, nagios, netflow, ids/ips etc. BSD, Linux,

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

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