GRANDSTREAM RELEASES NEW 12-LINE ENTERPRISE IP PHONE FOR BETA TE

GRANDSTREAM RELEASES NEW 12-LINE ENTERPRISE IP PHONE FOR BETA TESTING

New Beta test features a desktop IP Phone ideal for enterprises and high call volume users

Boston, MA (November 5, 2015) – Grandstream Networks, connecting the world since 2002 with award-winning SIP unified communication solutions, is releasing its next enterprise IP phone for beta testing by the Grandstream Beta Club. Grandstream will be opening applications for the beta test of a new color-screen enterprise IP phone from November 5 – 17, 2015. Go to betaclub.grandstream.com to apply.

Grandstream will introduce this new phone to its award-winning GXP21xx series of color screen enterprise-grade Gigabit IP phones which currently includes the GXP2130, GXP2140, and GXP2160. This new 12-line phone, or the GXP2170, further enhances the Grandstream solution, providing a wide variety of products for a complete IP unified communications package.

For this beta test, the Grandstream Beta Club is looking for IP telephony enthusiasts; those who like to experiment with and test new products, and those whose innovative ideas can transform the way in which IP telephony is applied in today’s market. Ideal applicants are those who have prior experience with IP phones and IP phone networks. Those who display a dedicated interest in exploring IP technology are also invited to apply.

The newest product being released to the Grandstream Beta Club is a high capacity IP phone ideal for enterprise and high call volume use. This Linux-based model (GXP2170) features 12 line keys with up to 6 SIP accounts, 5 XML programmable soft-keys, 48 digitally programmable BLF/fast-dial keys with dual color LED, and the ability to use the GXP2200EXT extension module for up to 160 additional BLF/fast-dial keys.

This leading-edge phone also features a vivid 4.3” 480×272 color LCD, 5-way conferencing, HD wideband audio, full-duplex speakerphone with advanced acoustic echo cancellation, dual Gigabit network ports, integrated PoE, Bluetooth, USB and EHS. Fifty (50) units of this phone will be available for shipping to selected Beta Club members by November 20, 2015.

Program Availability & Pricing

Each product beta test released through the Grandstream Beta Club will have a limited amount of beta test units available. Application to the Grandstream Beta Club is open to anyone and is FREE. Grandstream will select participants based on applicant qualification. The cost to participate in a beta test will be $1 to cover handling; however, the product model and shipping will be free of charge for each active beta tester. There is no charge to applicants who are not selected for a Beta test.

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5 Ways You Can Successfully Implement SIP

Voice over IP (VoIP) uses Session Initiation Protocol (SIP) to convert voice to data and transmit those data over public or private IP networks. Businesses have used VoIP as a way to cut costs, add features and capabilities, and make more efficient use of their IP networks. Here are five methods you can research and use to make this process even better.

1. Service Provider

Find out if your service provider owns its own physical delivery methods for secure end-to-end service. Carriers that patch multiple networks together (some of which may not include SIP) may cause issues in quality of service. When troubleshooting multiple networks, it can take longer to diagnose issues and delay issue resolution. Carriers that have a native SIP network are ideal because such a network is a more efficient way to deliver IP traffic.

2. Advanced Features

To gain the most out of SIP, ask your service provider if it has a business continuity and disaster recovery plan. Your provider should have a failsafe plan in the event of outages on its end to ensure that emergency calls go through at any time. No service provider should ever charge for E-911 service. When properly managed, local emergency dispatchers should be able to find a caller’s location in the same amount of time as a 911 call.

3. Convergence

When researching a VoIP provider, ensure that your IT engineers and service provider can work together to design an efficient SIP network. Most service providers already have a plan or have experience working with companies that have on-premises data networkers. Each solution may be unique based on the needs of the individual business, but many service providers are more than well equipped to handle this. If you plan to grow your business or have already experienced significant growth, your telephony plan should be scalable to meet your business needs.

4. Security

When securing a VoIP solution, firewalls and other devices should be sufficient to provide adequate security. Some networks use a session border controller (SBC) device for traffic control, additional security, and configuration. If you use both a firewall and an SBC, your firewall should be placed outside your network as a first line of defense for all network traffic. Your service provider should offer SIP registration on the SIP trunk.

5. Redundancy

You have the option to centralize or distribute SIP trunking. If you have a large network (more than 500 session instances), redundancy is required. Fewer than 500 sessions are a good idea but may not be required. An assessment of the impact of the event of an SIP trunk failure to a nearby backup location until service resumes may be necessary. Your service provider will be able to provide the details on its end. If an SIP trunk goes down, outbound calls should be routed seamlessly through a traditional public switched telephone network gateway. Inbound traffic alternates should be offered through any service provider.

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Sangoma Vega50 VS0119 - 4 FXO Voip FXO Gateway

Sangoma Vega50 VS0119 - 4 FXO Voip FXO Gateway

The SangomaVega 50 media gateway connects legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones to IP networks.

• 2 – 10 Ports, analog or BRI
• Voice, FAX and Modem Support
• Sangoma Resiliency Enablement Suite (SRES)
• Browser-Based GUI Interface for quick and easy configuration
• Voice, FAX and Modem Support
• Sangoma Network Appliance Provisioning (SNAP) tool
• Flexible Call Routing for Fallback and Least Cost Routing
• Emergency PSTN Backup
• SIP and H.323 support

The Sangoma Vega 50 VoIP Gateway creates a connection between any Analog or BRI device and a VoIP Network.  The Vega 50 can be factory configured to support a variety of configurations.  When configured for analog telephony, the Vega 50 can support up to eight FXO ports or eight FXS ports with 2 FXO ports.  When configured for BRI/ISDN2, the Vega 50 can support up to 4 BRI interfaces. Each BRI interface can be independently configured as network side or terminal side. Therefore, the Vega 50 media gateway can be connected to both a PBX & the ISDN simultaneously. The GUI and Web-based SNAP Tool also make the Vega 50 one of the easiest gateways to provision and deploy.

The Vega 50 is an outstanding choice for a variety of Enterprise and Service Provider Application:

Enterprise Applications:

• Resilient Branch Office Support for Hosted or Premise Based PBX or IP-PBX
• FXS endpoints and Emergency Number calling for Hosted or Premised Based IP-PBX
• Support for, door entry phones, POS, elevator/lift phones

Service Provider Applications:

• Customer premises gateway for SIP trunking
• FXS endpoints and Emergency Number calling for Hosted IP-PBX

Analog

• 4 FXS + 2 FXO
• 8 FXS + 2 FXO
• 4 FXO
• 8 FXO

BRI / ISDN2

• 2 BRI interfaces (4 channels)
• 4 BRI interfaces (8 channels)
• 8 BRI interfaces (16 Channels, subject to MOQ from customer)

Each BRI interface can be independently configured for connection to the PSTN or to a PBX. The Vega 50 media gatewaycan, therefore, be connected to both a PBX & the ISDN simultaneously. This configuration provides:

• No disruption to the configuration of existing equipment
• Flexibility & choice for call routing

Interfaces

VoIP

• SIP
• H.323 version 4
• Audio codecs:
  • G.711 (a-law/µ-law) (64 kbps)
  • G.729a (8kbps)
  • G.723.1 (5.3/6.4 kbps)
  • G,726
  • Clearmode
  • GSM (Optional)
• FAX Support – up to G3 FAX, using T.38
• Modem Support – up to V.90, using G.711
• Up to 8 VoIP channels

Telephony Interfaces

ETSI BRI

• 2 or 4 S/T interfaces presented on RJ45
• Point to point or point to multipoint
• Each interface can be configured NT or TE

Analog

• 4 or 8 FXS ports with 2 FXO ports presented on RJ11
• 4 or 8 FXO ports presented on RJ11
• 600R, 900R or CTR-21 line impedance

LAN Interfaces

• 2 RJ-45s, 10 BaseT / 100 BaseTX, full / half duplex

Features

Telephony Features

Available on FXS ports:

• Call Waiting
• Call Forward   Unconditional, Busy, No-Answer
• Call Transfer   Blind, Consultative
• 3 Way Conference
• Do Not Disturb
• Message Waiting Indicator – Audible, Visual
• Music on Hold
• Executive Barge

Identification

• Caller ID presentation
• Caller ID screening allows connections to be accepted only from selected call sources
• H.323 gatekeeper registration
• SIP Registration & Digest Authentication

Operations, Maintenance & Billing

• HTTP(S) web server
• Radius Accounting & Login
• Remote firmware upgrade:
  • Auto code upgrade
  • Auto configuration upgrade
• SNMP V1, V2 & V3
• Syslog
• TFTP / FTP support
• VT100 – RS232
• Telnet/SSH
• Voice readback of IP parameters

Routing & Numbering

• Dial Planner – sophisticated call routing capabilities, standalone or gatekeeper/proxy integration
• Direct Dialing In (DDI)
• SIP registration to multiple proxies
• FXS ports support Call Waiting & Call transfer (blind & consultative)
• NAT traversal

Call Quality

• Adaptive jitter removal
• Comfort noise generation
• Silence suppression
• 802.1p/Q VLAN tagging
• Differentiated Services (DiffServ)
• Type of Service (ToS)
• QoS statistics reporting
• Echo cancellation (G.168 up to 128ms)

Security & Encryption

• Media – SRTP (Optional)
• SIP – TLS (Optional)
• Management – HTTPS, SSH Telnet
• Configurable user login passwords
• Enhanced Network Proxy (ENP) (Optional)

Hardware

Certification

EMC (Class A)

• EN55022
• EN55024
• FCC Part 15
• AS/NZS3548
• VCCI

Safety

• EN60950
• IEC60950
• UL60950
• AS/NZS60950

Telecoms

• TBR21 (FXO)
• FCC Part 68
• CS-03
• TBR3 (BRI)

Environmental

• 0° .. 40°C
• 0% .. 90% humidity (non-condensing)

FXS Line Length

• 1500ohm, up to 8km

Indicators

LED:

• LAN: speed / activity

Physical Dimensions

• 299.73mm (11.80″) x 44.45mm (1.75″) x 237.06mm (9.33″) width/height/depth
• Weight: 1.02kg (unit only)
• Rackmount brackets supplied

Power

• Standard product: External 12V 24W AC/DC adapter

Program Storage

• Code & configuration data are stored in FLASH & executed from RAM

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