This is an exciting feature although the first go-round that Ubiquiti had with their line up wasn’t well received and they didn’t deliver on what the people expected. Now, they are back at it with the 2nd generation line-up of devices. We’ll see how they fare.

There is already a competitive phone in the new lineup for $29 each! That’s cheap! It’s the UVP-FLEX. It’s a no-frills looking phone but it does everything that you should need. Original purchase price was $49 but has come down. So far, this device doesn’t let you program your own custom SIP provider into it so you’ll be limited to UniFi as a PBX/VOIP system.

Just the same as the first generation and second generation UVP phones the phones with the touch interfaces still have the ability to have the SIP provider configured manually on them.

Even with the hardware you still need the controller software “Unifi Talk”

To get Unifi Talk working you will need a “Cloud Key Gen2 Plus” or an “Unifi Dream Machine Pro” a.k.a CKG2+ and UDM-PRO.

Then, you’ll need to manually get a firmware version recent enough to include the installers for UniFi Talk. For my lab I just used the latest firmware version for my CKG2+.

For my lab I need to test if this is a viable replacement for a business phone system. Therefore, I need standard advanced calling features and the ability to port pre-existing numbers to the service.

From the Add Number web GUI there isn’t an option to port in external numbers from a 3rd party.

However, you can add a 3rd party SIP provider. If your number is not already a SIP trunk then it could be ported to a 3rd party SIP trunk provider. Then, the 3rd party SIP provider could be added to the UniFi Talk controller using the advanced settings.

Here is a link to the Ubiquiti support page that demonstrates how to add a 3rd party SIP provider.

So, in theory it could work. I have yet to test it.

The other thing that some people had written online about it was that they were unable to have 2 phones use the same outbound number when not in use. It seems that a phone number must be assigned to each phone to allow outbound calling. If there is no solution for this then it is not a viable business PBX alternative. I hope a firmware update resolves this issue.

**UPDATE** 12/24/2020

3rd Party SIP Trunk Provider (One that can port an existing number)

I decided on going with ClearlyIP as a SIP trunk provider for my 3rd party test. Mostly due to being a fan of the “Crosstalk Solutions” channel on Youtube. That guy does the world a huge favor sharing his knowledge and experience and a huge thank you to him and his channel. For the same reason Sangoma was another option but I liked ClearlyIP more mostly due to branding.


SIP Trunk (You need only 1)

I called ClearlyIP and asked for a “Calling Path” (that’s their nomenclature for a SIP channel). In essence, the SIP trunk (infinite calling paths) is $9.00 which is owned by the ITSP (Internet Telephony Service Provider) but you are charged per SIP channel on a usage basis of $0.095/minute. So, let’s consider the $9/mo. cost keeping the lights on.

Lines vs. SIP Channels (total simultaneous calls)

Consider your SIP channels ($17.99) your Inbound/Outbound Lines. You want the same number of SIP channels as Inbound/Outbound calls you expect to ever simultaneously occur.

Free Trial: ClearlyIP has a free trial option that give you the opportunity to test out the SIP trunk. You need a credit card on file but you get a $10 credit and are charged on usage @ $0.095/minute.

The sales person that I spoke to suggested that I create 5 channels to test it out. There is no harm to this as you only pay for what you use. You could set up 15 but if you’re just using one phone that won’t really affect your bill as your total aggregate calls are what you pay for.

DIDs – direct inward dialing (Line Numbers)

These numbers are essentially direct lines to a specific phone. Similar to an extension but you immediately skip all the calling tress, call queues, auto-attendants, etc and go straight to a specific person’s phone. This is great to reduce the number of calls that your front desk may have to otherwise pick up just to forward the call to you.

Regarding DIDs what you don’t need a DID for would be any phone that is meant to be called as part of a calling tree, or hunt group, directed by your front desk or an auto-attendant.

What you do need DIDs for is your published numbers (Main Line Numbers & Fax Numbers). Or, for users who need to have customers skip the calling tree to call you directly.


  • CKG2+
  • 802.3 af network switch
  • UniFi Talk Controller (Latest UniFi Firmware includes Talk)
  • ClearlyIP SIP Trunk (Free Trial)
  • 5 SIP Channels (5 calling paths in ClearlyIP language)
  • 1 DID @ $1.00 / month
  • ROUTER (THIS MATTERS ALOT!!!) – It’s important that SIP calls aren’t suffering from DPI (Deep packet inspection) or just having the ports that it needs to communicate blocked. You could be able to place calls but wouldn’t have any audio or garbled/digitized sounding audio. QOS (quality of service) settings also matter regarding traffic priority. You’ll want to hear what someone is saying more than have a network packet from windows update making it to your PC more timely than hearing your phone call. UPnP is the easiest configuration option for UniFi Talk.

Getting UniFi Talk to work with ClearlyIP SIP Trunk

Firstly, UniFi Talk is in Beta so will it even work? To answer that it is important to understand some of the inner-workings of UniFi talk that really isn’t published anywhere from what I found.

Well, UniFi talk is built on a well-known platform (FreeSwitch) which “is” compatible with ClearlyIP. I noticed this in the small text showing me that I could access the FreeSwitch CLI

When ClearlyIP set me up with a trunk I was provided with a

  • SIP Username
  • SIP Password
  • Keycode (Only useful for FreePBX or ClearlyIP appliances for autoconfiguration)

Irritatingly, my not-so-welcome packet didn’t include information such as the server address. This I had to find on my own. From the above link I found the server address in the .xml file example:

Regarding the built-in SIP trunks and channels that can be purchased through the web GUI the purchase and acquisition of those is handled by a Twilio API which means that Twilio is the ITSP for those numbers.

To add the information to UniFi Talk you have to go into the settings, add another provider and then create the fields that SIP trunk provider needs to authenticate you and assign you a DID.

At this point in the Beta I would recommend going with FreePBX(asterisk), PoE phones, and a Grandstream ATA if you need a commercial telephone solution. You could use a PCIe ATA card instead for neatness.

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