IP-PBX appliance GrandStream UCM6102

Functional overview and testing results

Evgeny Anvaer, Vladimir Dudchenko

SoftBCom, Ltd. (www.softbcom.ru) 27.02.2014

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.

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.

• 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

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.

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

Grandstream UCM6102 administering

Grandstream UCM6102 is managed through Web GUI

Grandstream UCM6102 administering

General menu appearance (with Russian language setting)

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.

IAX –trunk creation

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.

Connecting to phone operator by PSTN

Connecting IP PBX to analog phones

8. FXS–ports do not require any special settings. Just create Extension after phone has been connected.

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.

Firewall settings

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».

NAT Settings

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

Queue creation

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

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.

Backup copy creation

Backup administering

IVR creation

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

IVR creation

Defining functions for time-out and wrong option events.

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

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).

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».