similar to: Disappearing voicemail?

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.

Hey, all. I've got a client who's interested in possibly using a softphone for his receptionists. While I've certainly used some softphones for single extensions, I'm not sure which one I'd suggest for a receptionist. Any favorites? Thanks, -Ken

So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local & long distance for free. It also does 'internet'. 'Just give me

Hi Running bristuffed 0.3.0-PRE-1f which is 1.2.1. Using *2 in features.conf for attended transfer. Works well if someone answers. But the following sequence causes issue: 1. Receptionist takes call. 2. *2 then 123 to transfer to extension 123. 3. 123 is busy or does not answer so receptionist hears 123 voicemail 4. How can receptionist reconnect to calling user - could wait for voicemail to

Dear all, Uwe e-mailed me yesterday about the problems some time ago. I had hoped someone would have a look at our computers, but due to some unfortunate circumstances, it hasn't happened. However, it seems unlikely that it is connected to a specific piece of hardware. It could be an architecture-dependent problem though. I also had a word with those who did the installation of R, and they

Hi there, I would like to setup up my Asterisk to do this: receptionist answers the call, caller says he wants to leave a voicemail message for Ashleigh, receptionist transfers the call to Ashleigh's voicemail I guess It has something to do with dynamic features, or probably blind transfer to special ext. might do it what would you recommend? cheers!

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

I got dragged away from Asterisk (somebody made me an offer I couldn't refuse for system administration), but I'm thinking about seeing if I can't get it deployed at my new employer. Regardless, there are two things about older voicemail that used to annoy me: - Dial by name. Has anyone made it so it can be first or last? - Jump to voicemail; you used to have to actually dial the

Asterisk 1.2, Fedora Core 4: When I leave a voicemail message, it writes the necessary files to the INBOX: msg0000.gsm msg0000.txt msg0000.wav msg0000.WAV When I hang up, the files are erased. There is no indication of anything untoward in the logs: -- x=0, open writing: [...]/INBOX/msg0000 format: wav49, 0x99ce778 -- x=1, open writing: [...]/INBOX/msg0000 format:

Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy

Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So

I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and

I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long

I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? For example, if someone dials 1000 to check voicemail at site A. The dialplan will be something like this on Site A: [context-for-phones-at-one-location] exten =>

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

hello all- Long time listener, first time caller. This is a great list and has given me tons of help as I've set up * for the first time. I've got an asterisk system up and running at a new company, and it does about 99% of what we need it to do. TelephonyWare has been our equipment supplier, and has been great with support, but I've got an issue that has us both stumped.

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

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

Hey all... Scenario (INTERNAL) 1 Call comes in to receptionist and gets transferred to someone 2 No one picks up that transfer 3 Call goes back to receptionist Now when the call goes back to the receptionist, how can I change either the ringer, the callerID or both? * TIA