similar to: Mitel 3300

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

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

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---

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

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, 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

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.

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

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>

Anyone in Winnipeg Canada?

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

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

Does anyone know if this phone can or does work with Asterisk ? I have a potential client who wants to throw away their current mitel ip system (keeps crashing on a regular basis - not good for a phone-based company), but are reluctant to purchase new handsets. Thanks. Julian.

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

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

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

We are attempting to configure SIP trunking between asterisk 1.2.22 and a Mitel 3300 CX box. The Mitel machine will gateway to the PSTN for us. I found this earlier post about doing this from July: http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html Unfortunately the promised configs never came ;(. We're having the exact reverse problem: we can register with the Mitel