similar to: Read Back Caller ID Using Number Announcement in Digital Receptionist

Hi I make a Digital Receptionist ,but how can I attach it to an extension Help regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic

Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From

Hi to all, I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? One SIP account, has a limit on concurrent calls? I saw that the SNOM-360 can handle up to eleven SIP accounts. Thanks -- .:FaberK:.

I've tried searching the archives but come up empty. I'm looking for a "receptionist" application, that could be used in a business center. Lots of different incoming phone numbers (200+), all need to be answered in a different way, by a human. This is typically enabled with a terminal of some form that pops up the company and contact information for the called number, and a

Newbie question alert! I was just wondering how a receptionist phone would work with Asterisk. (I've never had a real job, so I've never really looked at different phone systems). I have looked around on the internet and seen that you can purchase 24 line phones; how does that get connected? What kind of wiring goes to the reception phone? How would I go about adding one of these to an

Hello, Can anyone provide me with an example of how they have set up their dialplan and handset for a receptionist desk that handles a large volume of concurrent calls? I'm having a problem with transferring calls while several calls are either answered or coming into a receptionist's telephone at the same time. Thanks, Colin -------------- next part -------------- An HTML attachment

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

I am so far unable to get the busy lamps on a Snom 220 to work either with current cvs or asterisk 1.0. I am using the hint extension and the Snom 220 just as described in the "mini-howto" on: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html There are also a couple of wiki pages referencing this: http://www.voip-info.org/wiki-Asterisk+standard+extensions

Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to,

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work

Hello list, I would like to share a little problem with you : I've lost an 5Gb partition in an installation of Ututo GNU/Linux. It seems that the installation script does not took care of my /home and it tried of format it. When I saw that my old sda8 partition was being formatted as ext3 (fast format) I immediately, hard-restart my computer. After a run level 1 restart, I discovered that it

I am looking at placing a system in an office with a central receptionist, and phones for each individual employee thereafter. Could I use a Snom 220 with additional keypads to view if the lines are in use by the other employees? Fred is in sales... A call comes into the receptionist and they transfer the call to Fred. The receptionist can tell Fred is still on the phone by viewing the assigned

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

Hi, Quick question. With an old phone system a receptionist receiving a call has 1 button to push to transfer calls to a specific extension, with Asterisk, a receptionist would actually put the caller on hold, pick up another line, call the extension, ask if the person is available, hang up pick up the caller again and transfer. To me it's seems a long way to simply do a receptionist

I have two contexts and there I have some sip clients and some iax clients, in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax clients I have some extentions like 2000, 2001, 2002, 2003, etc. My extention is 2003 when I make a call the manager program show me that the extention 20 was originating the call (in red, and show my name/extention)! When I make a call to

I read this story and thought of Allison's prompt to "try not to think about blue eyed polar bears". Will she be banned from foreign travel now? Steve Prior -- snip -- WASHINGTON (Reuters) - Polar bears, sea ice and global warming are taboo subjects, at least in public, for some U.S. scientists attending meetings abroad, environmental groups and a top federal wildlife official

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

I'm sending in a set of voiceover requests to Allison Smith this afternoon. I haven't kept up with the -users list to know if there is someone keeping track of this stuff any more... We only have a few phrases for her to record, and if anyone has applications which require Allison's voice for the "asterisk-sounds" repository, let me know. I'll be sending this in