Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: pakman on May 04, 2017, 08:15:35 AM
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Currently, I have 30 sites each with their own PBX using analog lines and a few PRIs. I would like to centralize our system into one virtual PBX and start getting rid of analog lines and start utilizing SIP.
Wanted to find out what others thought about getting rid of analog lines and using SIP only except for fax lines. I have not used SIP before and would like to know what kind of pitfalls I might run into switching over to this technology. Bandwidth between all of our sites is good most have 10 meg fiber and a few Ethernet over coax running 5*5.
Thanks
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As long as you have the bandwidth I think that's the way the industry is moving. For the fax lines I've gotten good feedback regarding webfax products where you receive faxes in an online box and send faxes via a virtual printer on the PC. If someone wants to send a fax that's not already on their PC they can scan it and then fax it that way.
Depending on how widespread your sites are you might want to verify number portability with the carrier you select for the SIP trunks.
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We do very little TDM stuff on larger (100+ phones) installations anymore, it's all SIP now and we as long as you have reliable connections to your ISP and/or SIP provider it is great and trouble free. Converted a lot of customers over from PRI or analog lines to SIP to. My suggestion is to pick a quality carrier over the cheapest ones though, it makes a difference.
For that matter, we rarely do even do physical equipment anymore... everything is virtualized now days.
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... everything is virtualized now days.
I agree, but for standalone sites the installations with CX are still much cheaper.
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... everything is virtualized now days.
I agree, but for standalone sites the installations with CX are still much cheaper.
If they don't have a virtual environment, then yes I agree... Or if there is a need for lots of analog devices like assisted living facilities an AX makes sense.
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Agree, and another reason is that for standalone virtual installations enterprise licenses have to be used :-(
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Thanks for the feedback!
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I'm not sure if you they do secure calls on your sites using STEs, or STU III.
This sort of was our "downfall" after a number of our sites converted to SIP trunking. We had no choice but to reinstate analog trunks just for secure calls.
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I have not used SIP before and would like to know what kind of pitfalls I might run into switching over to this technology.
If you use only MCD -> provider SIP gateway, be aware that the MCD will not be in the media path [I think MCD is unique among phone systems in this regard]. You will need to add your own SIP gateway to anchor the media [eg MBG] if your network requires it.
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Thank you.
I recently found out we will need a MBG for this to work.
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There may be some things to consider when going from TDM to SIP. One would be license costs and if an MBG is involved thats 2 to license. I think that in many cases, an Adtran 908e would do the conversion at a better price point.
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How much do they cost? They're not obviously available in EU.
I have been building VMs with FusionPBX on to handle media anchoring but not every site has VM infrastructure I can put it on.
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I have been building VMs with FusionPBX on to handle media anchoring but not every site has VM infrastructure I can put it on.
I'm interested in this, I have been playing with FusionPBX but not extensively. I installed it on an Intel NUC on a small site that had some issues.
How do you use it?
Do you have the FusionPBX register to the provider and just route calls from the FusionPBX to the Mitel as a B2BUA? or do you use it as some sort of outbound proxy.
I also recently came across www.sipwise.org which I haven't tried but it looks interesting.
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I *think* it's a B2BUA but I'm not 100% on the terminology. There's no registration involved at all, the provider we use does everything as "IP authenticated". I've basically configured FusionPBX in such a way that it mimics how the carrier behaves, so I can easily flip carrier SIP trunk back to the Mitel with the minimum of config changes. The call routing in each direction just chucks everything that comes in back out of the other SIP trunk.
CDR logs and call blocking features are very handy.
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B2BUA (Back to Back User Agent) is involved in the call (unlike a 3300), as you said, "media anchoring"
I had Fusion register to the provider as I had a MiVO250 that was not keeping registration as their link was unstable. The FusionPBX would just keep trying.
The the MiVO250 was "IP Authenticated" to the FusionPBX and I just routed calls in and out. Worked well.
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My apologies for the late reply on your topic, but you have a few options to consider. One could be moving stand alone sites to a hosted VoIP platform, however the downside to this is that if you lose internet you lose voice. If you choose to go with hardware, like an MCD, you'll also need to get additional hardware such as a SIParator (Ingate). This acts as your MBG and can be configured for of your SIP needs, including codecs, etc. but like with anything else requires licensing. Depending on how many concurrent calls you'll need to support, you'll need to make sure to have the proper dedicated bandwidth. For Example, on a 10Mb connection you could handle up to 80 (80%, industry standard) concurrent calls. This equates to purchase 80 SIP trunks, then dividing them up as Session & Usage accordingly. You can also go with a cloud base platform, like Cloud PBX w/PSTN Calling if you wanted to integrate this with SFB. With this option you move away from purchase desk style phones, but the options are they if say "older" employees want them.