Grandstream UCM6102 is an Aterisk based IP PBX appliance. Perfect solution for SMB heaving affordable price ($440 or so), it supports up to 30 concurrent calls. But you should check carefully if it really fits to your needs, to avoid disappointments and unjustified money spending.

1. IP-PBX appliance GrandStream UCM6102
Functional overview
and testing results
Evgeny Anvaer, Vladimir Dudchenko
SoftBCom, Ltd. (www.softbcom.ru)
27.02.2014

2. Grandstream UCM6102- General features
GrandStream UCM6102 is an IP-PBX appliance introduced in July last year.
It’s intended for small-sized companies and has affordable price.

3. Grandstream UCM6102- General features
• The IP-PBX embedded software is
Asterisk 1.8.23.1. It’s a well-known, reliable
and approved version of Asterisk, keeping a
good balance of advanced functionality
and stability.
• Grandstream UCM6102 is very attractive
for its potential users – having affordable
price (one can buy it for $440) it has a set
very advanced and promising features.

4. • Grandstream UCM6102 is a modern
solution intended for replacing legacy
communication office system in small
and mid-size companies.
• But just the same it has a few
features which should be taken into
account to avoid disappointment and
unnecessary expenses. We just point
out such features.
• We appreciate help of VOIP GROUP
which provided IP-PBX Grandstream
UCM6102 for this testing.
Grandstream UCM6102- General features

5. Grandstream UCM6102 connections
1. The PBX has 2 FXO and 2 FXS analog ports. In case of powering the device off the
FXO lines get directly connected to FXS lines. It’s a specific way of fault tolerance
implementation. FXS ports can be connected to analog fax machines.
Grandstream UCM6102 has 2 Ethernet ports: WAN and LAN.

6. Grandstream UCM6102 connections
2. Grandstream UCM6102 has 2
Ethernet ports: WAN and LAN.
It also features embedded NAT
and firewall. Thus ISP’ line
could be connected directly
without any additional router
or firewall.
3. In case of E1 flow, SIP/E1
gateway is to be implemented.
But as it has almost the same
price as the PBX itself, use of
this appliance in such a case is
not reasonable, while due to
our information the vendor
plans to include E1 port into
the next versions of the
product.
General features
• 30 concurrent calls
• 2 FXO, 2 FXS ports
• 2 x RJ45 10/100/1000 (PoE)
• 2 fault tlerant PSTN
• USB, SD
• 3 virtual conference rooms
• 4 Gb flash memory
• Voice codecs: G.711 A-law/U-law,
G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B,
iLBC, GSM
• Video codecs: H.264, H.263, H263+
• Network protocols: TCP/UDP/IP,
RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP,
TFTP, SSH, HTTP/HTTPS, PPPoE, SIP
(RFC3261), STUN, SRTP, TLS
• Management: TFTP/HTTP/HTTPS
• Dimensions: 226mm (L)x 155mm
(W)x34.5mm（H)
• Power supply - Input: 100-240v, 50-60Hz,
output: 12 v (DC), 1.5A

7. Grandstream UCM6102 administering
Grandstream UCM6102 is managed through Web GUI

8. Grandstream UCM6102 administering
General menu appearance (with Russian language setting)

9. Grandstream UCM6102 specific features
administering and IAX trunk creation
4. The Asterisk configuration files in this PBX are not reachable by any way except using
Web GUI. The CLI options are not documented. The vendor states that the PBX
design doesn’t permit direct manipulations at OS or Asterisk internals level. So only
one VLAN is possible, and no direct connections of more than one SIP provider
applicable – in some cases it becomes a real limitation.
5. Connecting the PBX to remote Asterisk using native IAX2 protocol was tested by
connecting to separate FreePBX installation. For establishing such connection it’s
necessary to create IAX trunk in Grandstream UCM6102 and the same trunk in
remote Asterisk. Attention: the PBX names such trunk automatically (see it in the top
of the trunk creation screen below) and this ID shouldn’t be modified. To finalize the
process - create Inbound and Outbound Routes for both sides.

10. IAX –trunk creation

11. Connecting to phone operator
by SIP and by PSTN
5. For connecting the PBX to a phone operator by SIP protocol you need to create SIP
trunk as described above, using SIP type trunk instead of IAX2. For connecting the
PBX by analog port you need to create analog trunk as below in paragraph 7.
6. For connecting to telephone operator by analog line you need to use FXO ports.
When implementing the PBX in Russia you need to place between PSTN input and
the PBX FXO port a detector of hang-up signals. For each FXO connection create
analog trunk in the PBX. To make PBX capable to detect hang-up and congestion
signals automatically push a special button “Detect” in the trunk creation screen.

12. Connecting to phone operator by PSTN

13. Connecting IP PBX to analog phones
8. FXS–ports do not
require any special
settings. Just
create Extension
after phone has
been connected.

14. Fault tolerance and
WAN connection
9. In case of analog lines Grandstream UCM6102 is fault tolerant to power supply
issues. When the appliance is powered off the FXS-ports get directly connected
to FXO-ports and the phone communications keep running
10. The PBX can operate as a switch or as a router. So Grandstream UCM6102 can
be directly connected to ISP (see paragraph 4). Firewall and Fail2Ban settings
could be modified as shown in the next slides.

15. Firewall settings

16. Fail2Ban settings
11. By default
Fail2Ban is
disabled. With
Fail2Ban enabled
a white list of
reliable devices
could be set. All
blocked devices
will be unblocked
automatically
after the delay
established (in
seconds) in
«Banned
Duration».

17. NAT Settings

18. Queue creation
12. Queues can be
easily created
by GUI. The
method of
users log in/off
to the queue is
set separately.
Queue is
associated to
the specific
extension
number

19. Queue creation
Code settings for operators logging into the queue in and logging out.

20. Some specific functional features
13. Supposing the PBX could be used for a small call center we didn’t find any queue
monitoring tools. It’s not possible to involve external monitoring tools either
(e.g., Loway QueueMertics) due to closed embedded Asterisk configuration.
14. The PBX can record calls in wav-format and place records on internal disk of 4 GB
size. The calls could be also recorded to a flash card or SD-card in case of their
presence. One minute of call record has 948 KB volume.
15. Backup could be accomplished to internal disk, flash- or SD-card or external
SFTP-server.

21. Backup copy creation

22. Backup administering

23. IVR creation
16. IVR are
created by
GUI. In one-
level menu
the specific
functions get
bound to
buttons
pushed during
IVR execution.

24. IVR creation
Defining functions for
time-out and wrong
option events.

25. Call transfer and call forward
17. Any call can be
transferred to any
phone number
(internal or
external) during
talking by dialing
special feature code.
By default blind
transfer has ‘#1’
code and attended
transfer has ‘*2’
code. In this
example all calls to
5001 will be
unconditionally
forwarded to mobile
number with area
code 905

26. Feature Codes for users
18. There are no personal
settings of GUI for PBX
subscribers, but they
can use feature codes
(which are set by
administrator) to
accomplish necessary
functions. Feature
codes could be bound
to variety of functions:
call forward, call
parking, DND (Do Not
Disturb) etc.
Dialing such codes
subscribers can call
some specific functions
(The upper side of the
screenshot).

27. Outbound routes defined by rules
19. For each rule user
can define several
patterns, and all
outbound calls
complying to a
specific pattern will
use corresponding
trunk, which is also
defined in this rule.
In case of primary
trunk unavailability
or being busy a
failover trunk could
be used. For
creating failover
trunk click the
button «Click to add
failover trunk».