similar to: Mitel integration

Does anyone know anything about the Fidelio and Mitel protocol for hotel / motel? Are these industry standards or proprietary formats? Are there open standards for communication with Hotel management software's that could be used in conjunction with a custom asterisk deployment? Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Howdy, Has anyone worked on a Mitel-2000 emulation for PMS integration (Hotel mgmt systems)?  Hoping to get my hands on the protocol definition (RS-232!!) for check-in/check-out/housekeeping/CDR, but if someone has already done I would totally buy it. Cheers, -- Jeff LaCoursiere StratusTalk, Inc.

I have a problem which I need to resolve. We are trying to put an asterisk between a Mitel PBX and the world. We are adding Voip service via Asterisk. Here is are config files for the settings but our problem is the following. We are able to send calls to the Mitel pbx and it's the T1 connections is green saying it's ok. The support department from Mitel said that they use e&M and

Hi all, Our company has deployed a Mitel 3300 system (only about 2.5 years ago) and we are experimenting with setting up Asterisk in our head office (for business continuity, ie we have a bird flu epidemic and no-one can come in, therefore use SIP softphones at home to co-ordinate activity) and at a remote site in the Isle of Man (connected via 2Mbps SDSL) Ideally we'd like anyone on either

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Hi All, Our experience with * to date has been a bit limited. It's a 4xCisco 7960 network, linking our head office with a faraday caged datacenter. As a way of putting voicecomms into a sealed room, it was quick and easy to deploy, and works very well. As typically happens, we've now thought about extending the use of asterisk - and a new opportunity has cropped up. In three months

Hi, Some Mitel products, which are CentOS 6.x or 7.x servers based with some telephony services added, are only supported on VMware virtualisation platform, even all if theses CentOS guests are certified on last CentOS/Redhat Virtualization or Microsoft Hyper V platforms. So, at he moment we have three bad choises : - Migrate part of our virtualization services from KVM/libvirt to VMware ; -

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

I have a small government department that wants me to implement a Asterisk installation, however, they connect to the Government PBX, a Mitel SX200, and want to keep the ability to do that. I know there is no chance to connect the digital extension lines, but would it be possible to have the pbx admins send analogue extensions over and have those lines interface through an FXO interface? Or

Hello, I have seen the Mitel 5055 SIP phone mentioned a few times on the list, does anyone have any wonderful or horrible things to say about it? We are thinking about using them because they have many more programmable buttons than the Snom200 phones and are about $70 cheaper. Thanks, MATT---

Has anyone here had any experience with Mitel 5220 IP phones with Asterisk? Basic features are working good, but I'm looking for more advanced features like sending text to the display or having the lights on when an extension is in use via the hint subscription. Thanks.

Eric "ManxPower" Wieling wrote: >> >>> span=1,1,0,d4,ami >>> e&m=1-24 >>> > > Looks like you have told Asterisk to get it's timing from the Mitel. > I'll bet the Mitel is trying to get it's timing from Asterisk. > > Try span=1,0,0,d4,ami and run ztcfg -vvv > I just set this back. It was originally set to your

Good day everyone, I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and from extensions on both sides are completing successfully (details on config coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel 3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN calls through it successfully? Here is an extract of the log on Asterisk whenever I

Does anybody know if this is supposed to work and if so, what, if any, workaround is needed? I have other phones (Snom, Polycom) MWI working with this system fine. 6.0.0.19 (latest) Mitel SIP firmware is loaded. Thanks for your time, - Jesse -- Jesse Peterson <jesse.peterson@exbiblio.com>

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

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 Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can

Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice together? Mark Bergen Information Systems Manager Number TEN Architectural Group Winnipeg - 204.942.0981 Victoria - 250.360.2106 www.numberten.com<http://www.numberten.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: