Best SIP Desk Phones for FreePBX: Yealink vs Grandstream vs Poly

You’ve got FreePBX or plain Asterisk running, the extensions answer in a softphone, and now somebody wants a real handset on their desk. The best IP phone for FreePBX isn’t the one with the biggest screen. It’s the one your PBX can provision without a fight, that takes power from the switch, and that talks G.722 so calls stop sounding like 1998. That narrows the field to a handful of Yealink, Grandstream and Poly models, plus one analog adapter for the fax machine nobody will let you unplug.

Original content from computingforgeeks.com - post 180393

We ranked six SIP desk phones, one ATA and one DECT cordless kit on what matters once they’re plugged in: whether Sangoma’s Endpoint Manager has tested them, PoE and power supplies, wideband codecs and line keys. Specs come from Yealink, Grandstream and HP datasheets, and prices from live Amazon US listings. None of the hardware sat on our bench, so the PBX side was tested instead. We confirmed the PJSIP desk phone template below working on Asterisk 22.5.2 under Ubuntu 26.04.1 LTS in October 2026, with two softphones standing in for the handsets.

Pick the best IP phone for FreePBX by job

Read the EPM column first if you run FreePBX. “Tested” means the phone is on Sangoma’s commercial Endpoint Manager device list, so the PBX can generate its config file for you. A blank there does not mean the phone fails to register, only that you’ll provision it by hand or through the vendor’s own tool. Prices are Amazon US bands from October 2026, so check the live price before you buy.

Award	Phone	Price band	FreePBX EPM	Why it wins
Best overall	Yealink SIP-T53W	about $125	Tested	12 accounts, Wi-Fi and Bluetooth built in, gigabit
Best value	Yealink SIP-T33G	about $80	Not listed	Colour screen and gigabit for the price of a softphone headset
Best for BLF keys	Grandstream GRP2614	about $115	Tested	Second screen with 8 multi-purpose keys
Best premium	Yealink SIP-T57W	about $200	Tested	7-inch touch screen, 16 accounts
Best Poly	Poly Edge E220	about $125	Not listed	G.722 and Opus, Bluetooth 5.0, noise blocking
Best budget	Grandstream GRP2601P	under $40	Not listed	PoE phone for a lobby or a lab
Best ATA	Grandstream HT802 V2	about $40	Tested (HT-802)	Two analog ports and T.38 fax
Best cordless	Yealink W73P	about $150	Not listed	DECT base for up to 10 handsets

Three of the eight are on Sangoma’s tested list as named models, and the HT802 V2 rides on the HT-802 entry. That split decides more than the screen size does, because a FreePBX box with forty desks will spend far more of your time on provisioning than on any feature the phone advertises.

Check how we picked them

Every spec in this guide traces to a manufacturer page or datasheet fetched in October 2026, cross-checked against the live Amazon listing for the exact SKU. Where the two disagreed, we went with the manufacturer and called out the conflict, because two of the listings contradict the vendor on things you’ll notice (port speed and account count). The EPM column comes from Sangoma’s EPM supported devices page, last edited in September 2026.

We didn’t have these handsets in the lab, so we will not pretend to rate speakerphone quality or keypad feel. What we could test is the half of the connection you control, in this order:

1. Install Asterisk from the Ubuntu 26.04 packages on a lab VM.
2. Write one PJSIP template for desk phones and two extensions from it.
3. Register two baresip softphones as stand-ins for the handsets.
4. Place a call between them, then break registration on purpose with a wrong password, an unknown extension and a second device on one number.

That is where the codec, contact and authentication behavior in the later sections comes from, and it applies to any phone on this list, because the PBX sees a SIP user agent either way.

We left out Sangoma’s own phones to keep the comparison to the three brands in the title, and skipped anything only offered as a Renewed listing. A refurbished GRP2614 showed up at a lower price, but its title and condition field disagreed, which is a good reason to stay on the new unit.

Compare SIP desk phones for FreePBX at a glance

These are the columns that change a purchase for an Asterisk or FreePBX deployment. “No PSU” means the Amazon listing or the box contents say the power adapter is not included, so the phone needs PoE or a separately bought adapter.

Phone	SIP accounts	Display	Ethernet	Wireless	Power
Yealink SIP-T53W	12	3.7″ 360×160 mono	2 x gigabit	Wi-Fi + Bluetooth 4.2	PoE, no PSU
Yealink SIP-T33G	4	2.4″ 320×240 colour	2 x gigabit	None	PoE, no PSU
Grandstream GRP2614	6 (listing) to 12 (current page)	2.8″ colour + 2.4″ second	2 x gigabit	Bluetooth	PoE
Yealink SIP-T57W	16	7″ 800×480 touch	2 x gigabit	Wi-Fi + Bluetooth 4.2	PoE, no PSU
Poly Edge E220	16 lines on 4 keys	2.8″ 320×240 colour	2 x gigabit	Bluetooth 5.0, NFC	802.3af Class 3, no PSU in box
Grandstream GRP2601P	2	2.4″ 132×48 mono	2 x 10/100	None	PoE, no PSU
Grandstream HT802 V2	2 profiles, 2 FXS	None	1 x 10/100	None	Adapter (check listing)
Yealink W73P	10, up to 10 handsets	On the W73H handset	On the W70B base	DECT	Base powers the system

Two rows deserve a second look. The GRP2601P is the only desk phone here with 10/100 ports, so it’ll throttle a PC plugged into its pass-through port. And “no PSU” shows up on most of the Yealink listings, which is fine on a PoE switch and an annoying surprise on a home router with plain ports.

Start with the Yealink SIP-T53W for most desks

The T53W is the phone we’d put on most desks behind FreePBX. It’s on Sangoma’s tested list, it holds up to 12 SIP accounts, and Yealink builds dual-band Wi-Fi and Bluetooth 4.2 into the body instead of selling them as dongles.

Wi-Fi on a desk phone sounds like a gimmick until you have a desk with no cable run, and then it is the reason the phone works at all. The 3.7-inch screen is a 360×160 graphical LCD, not colour, but it is backlit with adjustable brightness. Its two gigabit ports pass a PC through at full speed, and the USB 2.0 port takes Yealink’s EXP50 expansion module when a receptionist outgrows the built-in keys.

Anyone who wants colour or a touch screen should look at the T57W instead. The Amazon listing we link (about $125, check the live price) says the power adapter is not included, so plan on PoE from the switch. A separate third-party listing bundles the adapter for a little more.

Save money with the Yealink SIP-T33G

The T33G gets you a colour screen, gigabit and four SIP accounts for about $80 (check the live price). For an office of people who mostly dial out, take calls and transfer them, there’s nothing here you’ll miss day to day.

The 2.4-inch display runs at 320×240, Yealink lists HD voice and EHS wireless headset support, and both Ethernet ports are gigabit, which the cheaper GRP2601P doesn’t match. What it leaves out is any radio: Yealink’s datasheet lists no Wi-Fi, no Bluetooth and no USB port, so headsets go through the RJ9 jack or an EHS cable. The same datasheet does list G.722 and Opus, so the cheap pick is still a wideband phone.

The catch for FreePBX is that the T33G is not on Sangoma’s EPM device list, while the older T31P is. It registers like any SIP phone, but you’ll either type the account in through its web UI or point it at a provisioning server you maintain. If you’re buying twenty, the T53W’s EPM support may pay back its higher price in saved hours.

Fill BLF keys with the Grandstream GRP2614

Pick the GRP2614 when someone needs to watch other extensions. Grandstream puts a 2.4-inch second screen beside the main 2.8-inch colour display, with 8 multi-purpose keys that page through 40 assignments and light up as digital busy-lamp fields.

It’s on Sangoma’s tested list, but read the EPM known limitations before you rely on it for the very feature you bought it for. Sangoma documents that Grandstream BLFs on line keys don’t work or update through EPM, which it blames on Grandstream firmware. In practice you’d let EPM handle the account and set the BLF keys in the phone’s own web UI or through Grandstream’s GDMS.

The spec sheet also shifted under the same model number. The Amazon listing (about $115, check the live price) says 6 SIP accounts, while Grandstream’s current page lists 12 accounts plus dual-band Wi-Fi 6 on hardware version 5.0 and later. Ask the seller which hardware revision ships before you count on the Wi-Fi. It is the wrong pick for a desk that will need an expansion module, because Grandstream’s spec table says it does not take one.

Put the Yealink SIP-T57W on a reception desk

The T57W is the T53W with a 7-inch 800×480 capacitive touch screen and 16 accounts instead of 12. It shares the same firmware family, the same built-in Wi-Fi and Bluetooth 4.2, and the same tested status in EPM, so it slots into a fleet of T53Ws without a second config template.

A big touch screen earns its price at a front desk, where one person watches a page of extensions, parks calls and transfers all day. On an engineer’s desk it is mostly a bigger clock. At about $200 (check the live price, and note the listing ships without a power adapter), you’d buy one or two of these and T53Ws for everyone else.

Go Poly with the Edge E220

The Edge E220 is the phone to buy when audio matters more than provisioning convenience. HP’s datasheet lists G.722, Opus, G.722.1, G.729AB and iLBC, plus Poly’s NoiseBlockAI on the speakerphone and Acoustic Fence on the handset. The noise handling is what sets it apart, because several cheaper phones here also list G.722 and Opus.

The rest of the E220 datasheet reads well too: a 2.8-inch 320×240 colour IPS screen, two gigabit ports, Bluetooth 5.0 with NFC, a USB-C port for headsets, and 4 line keys that page through 16 lines. It draws 802.3af Class 3 PoE at 13 W maximum, and the box contents list no power supply, so it is a PoE-only phone unless you buy an adapter.

Provisioning is where it costs you. Sangoma’s EPM page lists older Polycom VVX models (several as “Not Tested”) and does not list the Edge E series at all. Poly’s own answer is Zero Touch Provisioning and Poly Lens, or HTTPS provisioning from your own server, which works but is a second system to run. It sells for about $125 (check the live price).

Spend the least with the Grandstream GRP2601P

For a lobby, a warehouse wall or a homelab test extension, the GRP2601P does the job for under $40 (check the live price, stock was down to the last unit when we looked). You get 2 SIP accounts, a 2.4-inch 132×48 LCD, integrated PoE and EHS support for Plantronics, Jabra and Sennheiser headsets.

The Amazon title calls its ports gigabit. Grandstream’s spec table says dual switched 10/100, and we trust the manufacturer, so don’t hang a workstation off its PC port and expect gigabit file copies. It isn’t on Sangoma’s EPM list either (the GRP2612 and up are). Audio isn’t where it saves money: Grandstream’s GRP2601 datasheet lists G.722 wideband and Opus alongside G.711, G.729A/B, G.726, G.723 and iLBC.

Connect analog phones and fax with the HT802 V2

An ATA is how the old cordless phone, the door intercom or the fax machine joins FreePBX. The HT802 V2 has two FXS ports, so two analog devices each get their own SIP profile and extension, and it sells for about $40 (check the live price).

Fax is the reason to choose it over a cheaper adapter. Grandstream lists T.38 Group 3 fax relay up to 14.4 kbps with automatic fallback to G.711 pass-through, and TLS plus SRTP for the signalling and media. It has a single 10/100 Ethernet port, which is plenty for two analog lines.

Sangoma’s EPM list carries the HT-802 as tested but does not name the V2 revision, so check that your EPM template applies before you deploy a stack of them. T.38 also has to work end to end, which means your SIP trunk provider has to support it too. If it doesn’t, the HT802 V2 falls back to G.711 pass-through, which sends the fax as audio and is far less forgiving of packet loss.

Cut the cord with the Yealink W73P DECT system

Desk phones assume desks. For a shop floor, a clinic or a house, the W73P pairs a W70B base station with W73H handsets over DECT, and the base registers to FreePBX like any other SIP device.

Yealink rates one base for up to 10 handsets, 10 SIP accounts and 20 simultaneous calls, with up to 6 optional repeaters to stretch coverage. That’s a small office on one Ethernet drop. The kit we link is a base plus one handset at about $150 (check the live price), and extra W73H handsets are sold separately.

Leave it on the shelf if your phones never leave a desk, because a cordless handset needs charging and a DECT base adds a second layer of pairing to troubleshoot. It is also absent from Sangoma’s EPM list, so you’ll set the accounts through the base’s web UI or Yealink’s provisioning service.

Choose by provisioning, PoE and codec

Brand loyalty matters less than these checks. Work through them in order, because each one rules out phones before the next one is worth asking about:

1. Decide how the phone gets its config. FreePBX’s commercial EPM covers the models on Sangoma’s list, and Sangoma marks Yealink, Grandstream and Polycom as tested rather than certified, so EPM gives them basic provisioning only. Everything else goes through the phone’s web UI, the vendor’s cloud (Yealink RPS, Grandstream GDMS, Poly ZTP) or your own HTTP provisioning server.
2. Count PoE budget. Most listings here ship without a power adapter, and the E220 alone can pull 13 W, so size the switch before the phones. Our PoE switch guide covers budgets per port.
3. Match the pass-through port to the desk. A 10/100 phone like the GRP2601P caps the PC behind it, while every other desk phone here passes gigabit.
4. Put G.722 on both ends. Wideband only happens when the phone, the PBX endpoint and the far end all offer it, which our lab call showed below.
5. Check what’s still sold new. Polycom VVX units are cheap on the used market, but Sangoma lists most VVX models as “Not Tested”, and HP’s current Poly desk phone page leads with the Edge E series.

If you’re still deciding on the PBX itself, start with our FreePBX 17 install on Ubuntu or the Asterisk with FreePBX build on Debian 13, then come back for the phones. Plain Asterisk users can follow the Asterisk install on Ubuntu and Debian, which is where the next section picks up.

Register a desk phone to Asterisk with PJSIP

FreePBX writes its own PJSIP files, so FreePBX users create the extension in the GUI and skip this section. On plain Asterisk, one template keeps every desk phone consistent. We ran the following on Ubuntu 26.04.1 LTS with the distro’s Asterisk 22.5.2 package. The order is the point:

1. Define the transport and a desk phone template in pjsip.conf.
2. Create one endpoint, auth and AOR per extension from that template.
3. Add a dialplan context the extensions can call into.
4. Reload, point the phone at the PBX, and check the contact and the codec.

Our lab used the Asterisk package from the Ubuntu 26.04 archive rather than a source build:

sudo apt install -y asterisk

Open the PJSIP config next. On a fresh package install it’s a long commented sample, so keep a copy before you replace it. If you built Asterisk from our source guide instead, your file already holds a [global] section and NAT lines such as external_media_address, so add the templates and extensions below to it rather than replacing it, and skip the [transport-udp] section because your file already has one:

sudo cp /etc/asterisk/pjsip.conf /etc/asterisk/pjsip.conf.sample-orig
sudo vim /etc/asterisk/pjsip.conf

Replace the contents with this. The (!) sections are templates, and each extension inherits from them, so a codec change later is a single edit. Change both passwords to long random strings before any phone sees this PBX:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[desk-phone](!)
type=endpoint
context=from-internal
disallow=all
allow=g722,ulaw,alaw
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
dtmf_mode=rfc4733

[desk-aor](!)
type=aor
max_contacts=1
remove_existing=yes
qualify_frequency=30

[desk-auth](!)
type=auth
auth_type=userpass

[1001](desk-phone)
auth=1001
aors=1001
callerid="Front Desk" <1001>

[1001](desk-auth)
username=1001
password=Ext1001-ChangeMe

[1001](desk-aor)

[1002](desk-phone)
auth=1002
aors=1002
callerid="Workshop" <1002>

[1002](desk-auth)
username=1002
password=Ext1002-ChangeMe

[1002](desk-aor)

Listing g722 first is what makes calls wideband when the phone offers it, with ulaw and alaw as fallbacks. The rtp_symmetric, force_rport and rewrite_contact lines keep audio flowing when a phone sits behind NAT. With max_contacts=1 and remove_existing=yes, a new registration replaces the old one, which matters in the troubleshooting section below.

Next, give the extensions somewhere to dial. Back up and open the dialplan:

sudo cp /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.sample-orig
sudo vim /etc/asterisk/extensions.conf

This context lets 1001 and 1002 call each other and adds an echo test on *43, which is the quickest way to hear whether a new phone’s audio works in both directions:

[from-internal]
exten => _100X,1,Dial(PJSIP/${EXTEN},20)
 same => n,Hangup()

exten => *43,1,Answer()
 same => n,Echo()
 same => n,Hangup()

Load both files without restarting Asterisk. A reload is enough for new endpoints, but if you changed an existing transport, restart with sudo systemctl restart asterisk instead, because a reload does not rebind a running transport:

sudo asterisk -rx 'core reload'

Now set each phone’s SIP server to the PBX address (192.168.1.150 here), with the extension as both user name and auth ID, and the matching password. Once the phones register, list the contacts. Each extension should show one contact with status Avail and a round-trip time from the 30-second qualify:

sudo asterisk -rx 'pjsip show contacts'

Both of our stand-in phones came up on the first try. The baresip clients ran on the PBX VM itself, which is why every contact and log line below shows the PBX address:

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  1001/sip:[email protected]:507 31706eeeb3 Avail         0.346
  Contact:  1002/sip:[email protected]:507 56a5fe722e Avail         0.619

Place a call between the two extensions. From real handsets you’d just dial 1002 on the 1001 phone; we started ours from the Asterisk CLI, which rings 1001 first and connects it to 1002 when it answers:

sudo asterisk -rx 'channel originate PJSIP/1001 extension 1002@from-internal'

While the call is still up, check which codec the channel settled on. Your channel name will differ, so copy it from core show channels first:

sudo asterisk -rx 'core show channel PJSIP/1001-00000001' | grep -iE 'Format|State:|Connected Line ID:'

Both directions read g722, so this call is wideband end to end:

Connected Line ID: 1002
Eff. Connected Line ID: 1002
          State: Up (6)
  NativeFormats: (g722)
    WriteFormat: g722
     ReadFormat: g722
    State: sendrecv
    Formats: (g722|ulaw)

For packet loss and jitter on a live call, the per-channel RTP stats are more useful than any phone’s status screen:

sudo asterisk -rx 'pjsip show channelstats'

Twelve seconds into our call, each leg had sent and received about 611 packets with zero loss and jitter readings of 0.001. On a real LAN, a nonzero Lost column here points at the switch or the Wi-Fi link, not the phone model:

                                             ...........Receive......... .........Transmit..........
 BridgeId ChannelId ........ UpTime.. Codec.   Count    Lost Pct  Jitter   Count    Lost Pct  Jitter RTT....
 ===========================================================================================================

 7befb923 1001-00000001      00:00:12 g722      612       0    0   0.001    611       0    0   0.001   0.000
 7befb923 1002-00000002      00:00:12 g722      611       0    0   0.001    611       0    0   0.001   0.001

Before you expose port 5060 beyond the LAN, read our guide to securing Asterisk and FreePBX from VoIP fraud. Internet scanners probe open SIP ports constantly, and the passwords in this template are placeholders.

Fix a desk phone that won’t register

Most registration failures show up in /var/log/asterisk/messages.log with a clear reason, even when the phone’s screen only says “Registration failed”. We triggered each of these on purpose in the lab, and this pulls the two notices that matter out of the log:

sudo grep -E 'Failed to authenticate|No matching endpoint' /var/log/asterisk/messages.log

Each line carries a timestamp prefix in the file, trimmed from the excerpts below.

“Failed to authenticate”

A wrong password left the softphone stuck on 401 Unauthorized (Asterisk PBX 22.5.2~dfsg+~cs6.15.60671435-1), and Asterisk logged the reason with the source address and port. On a real network that address is how you tell a typo on one desk from a scanner hammering the PBX:

NOTICE[35702] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected]>' failed for '192.168.1.150:5079' (callid: b1a0960a44078515) - Failed to authenticate

Check that the phone’s auth ID matches the username in the auth section, not just the display name. Many phones keep a separate “Register Name” and “User Name” field, and only one of them goes into the digest.

“No matching endpoint found”

Registering as 1003, which does not exist, logged two lines with the same call ID. The first says Failed to authenticate, which sends people hunting for a password typo, and the useful reason is on the line after it:

NOTICE[35702] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected]>' failed for '192.168.1.150:5079' (callid: c62811c91ee09cbc) - Failed to authenticate
NOTICE[35702] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected]>' failed for '192.168.1.150:5079' (callid: c62811c91ee09cbc) - No matching endpoint found

The extension isn’t defined, or the reload did not pick it up. Run pjsip show endpoints and confirm the number is listed before you touch the phone.

A desk phone drops to Unavailable after a softphone logs in

This one surprises people. We logged a second client into 1001 with the right password, and with max_contacts=1 plus remove_existing=yes it replaced the first contact (the hash changed from 31706eeeb3 to 8066841d37). When that second client quit, 1001 sat at Unavailable, because the original device had no idea it had been replaced and stays off the PBX until it re-registers on its own timer. This is how sudo asterisk -rx 'pjsip show endpoints' reported it:

 Endpoint:  1001/1001                                            Unavailable   0 of inf
     InAuth:  1001/1001
        Aor:  1001                                               1

If someone wants a desk phone and a mobile softphone, give the softphone its own extension and credentials instead of reusing the desk phone’s, so neither one can bump the other. Raising max_contacts lets two devices share one AOR, but our dialplan’s Dial(PJSIP/${EXTEN}) only rings the first contact; the Asterisk docs use the PJSIP_DIAL_CONTACTS function to ring them all, which we did not test in this lab.

Calls connect but fall back to ulaw

Our first test call came up in narrowband, because the softphone’s codec list named G.722 at the wrong clock rate and it dropped the codec. The softphone logged audio codec not found: G722/8000/1 and answered with PCMU, so Asterisk honored what it was offered. Desk phones make the same mistake from a different direction: G.722 switched off in the phone’s codec menu, or moved below G.711. Fix the order on the phone, call *43, and check core show channel for g722 before you blame the handset’s audio.