similar to: Fidelio protocol and Mitel protocol

There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something

Hi, is geoRSS overlay supported with openstreetmap ? I can''t get it to work. Google maps georss http://blogs.intermedia.uib.no/motel/experiments/bergenGeoRSS.html Openstreetmap georss http://blogs.intermedia.uib.no/motel/experiments/bergenOSMGeoRSS.html As you can see, nothing shows in the openstreetmap example. Regards -- Rune Baggetun

While traveling, I found a motel with WiFi where Firefox and the DNS utilities, nslookup and dig, worked as expected, but wget, termtter (a CLI Twitter client), my Ruby on Rails applications, and several other programs could not resolve any domain names (getaddrinfo error, from memory). If I plugged the IP address obtained with nslookup into /etc/hosts, they worked as expected. Could anyone

Hi all, Looks like I'm going to make the trip over to Astricon next month, but finances being what they are and since I'll be paying for my own flight from the UK, I'm trying to cut down on costs. The problem I've got is that the hotel is about 4 miles from the nearest public transportation (Brookhaven station) and the shuttle bus is only for hotel residents. Hiring a car seems

All; I am looking for a solution a motel that is switching to VoIP. They are especially interested in prepaid services. That is, a resident will come to the office and pre-pay for phone services for their room. When the money runs out, service is shut off, and they need to come down to the office again. Accurate reports are very important. A search on voip-info.org shows several solutions out

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have

I am looking for a good analog desktop phone to use with asterisk and my sipura ATAs. I know I want Caller ID, MWI, a few programmable buttons (for asterisk features), and no external power supply (so my users can dial 911 through the SPA-3000 when the power is out). I spent some time looking at the phones at Fry's today, without finding exactly what I need. Do any of you have any

Anyone connected Asterisk to hospitality packages such as: Micros Fidelio Visual One Jonas We'd be interested in providing bounty on providing a connection to one or more (depending upon what the client selects) if our proposal goes through. Ultimately, about 300 to 600 stations will be provided. Ray. ------------------------------------------------- This mail sent through IMP:

I am working with a small inn (under 50 rooms) that is next to a ski resort. The inn just had Cat5e Homeruns to each room installed, with a patch panel in the basement. Now it's my job to connect each of the those rooms to the Internet. I think I have a Cisco switch that I can do Private VLANS with, however I thought of another solution. Has anybody seen or does anybody know of a VoIP

let's talk about writing tools that use markdown (or more broadly, another form of light-markup)... to begin with, there is the markdown "dingus"... from a demo perspective, it has been _fantastic_. it has provided interested observers a quick idea about the simplicity and the power of markdown. as a writing tool, it's a bit clunky; it will work, yes, but i'd doubt many

Your mention of the good ole' days of fidomail w/o spam got me digging for my Wildcat install disks. I found them both. Now I realize I got rid of my 5.25 inch drive in 1991. Hrmph. - btw I love the lyrics. :) <p><p>-----Original Message----- From: Robert Cole [mailto:robert@support4linux.com] Sent: Tuesday, July 16, 2002 14:16 To: vorbis@xiph.org Subject: [vorbis] Coffee

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hi all, Just a quick query; has anyone on here tried the Mitel 5340 IP Phone with Asterisk? If so, how did you find it - any problems, missing features etc? I've had a Google around and the general consensus seems good - I actually have a phone on its way to me early this week, but just thought I'd start investigating now ;) We have a live Asterisk 1.2 server, and an Asterisk 1.4 server

We are looking to find someone that is familiar with Fujitsu and Mitel PBX's. Email russc at inline.net off list. Thanks

Hi, folks: I've come into some Mitel 5020 IP phones. A client has made a significant investment in them and we want to see if we can use them in a new system. Are these even SIP sets? I haven't been able to find out. Mitel's site barely covers them (I was only able to find some user guides, which are effectively useless; they say nothing about configuration and provisioning). Has

I've configured an analog/single line off my mitel sx-2000 to a x100p card in my *, however when the remote caller drops the line I get a dialtone back from my pbx and the x100p doesn't detect the end of the call. Any way I can tell * to drop? I've tried call progress but it doesn't seem to work...

Hi, All! Hopefully someone can point me in the right direction. I have a Mitel SX2000 Light that I am interfacing with my Asterisk box over a T1. I can receive calls on the Asterisk box without any problems, but when I call an extension on the Mitel, I heard the ringing from the IP phone, but I don't see anything on the Mitel, other than channel 1 of the T1 showing as "Locked

I have a Mitel SX2000 with no voicemail. I'm wondering if it would be possible to use Asterisk to meet the need. This is a hotel property with mostly analog extensions. Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050628/25fd0ee3/attachment.htm

Anyone had any luck configuring the hold button successfully on a Mitel 5220 SIP phone? Xfer works great. Call completion works fine both directions. Audio has been beautiful. With a working system hold button, this would be a really usable phone. The hold button does make the call appear to be on hold: the light is blinking, the other party is still silently connected, I can pick up a call on

Rene, We are using the 5220 with Asterisk with no problems. The current firmware for the phone is 3.1.0.10, which you can download from http://sipdnld.mitel.com/ If you post your sip.conf file we can see if there's any errors. Cheers Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: