Cisco SPA502G

Cisco SPA502G

Cisco SPA502G supports the SPCP for working with the Cisco UC500 and SIPv2 protocol for working with the SPA9000 and other servers.
  • LCD (Black)
  • 2-port 10 / 100BASE-T RJ-45 Ethernet (IEEE 802.3)
  • The handset RJ-9
  • Built-in speaker and microphone
  • 2.5mm Headset Jack
Category:

Description

Cisco SPA502 Dubai

Cisco SPA502G is a reliable VoIP phone featuring both SIP and Cisco’s proprietary SPCP protocol. This phone supports a wide variety of call features and functions, including plenty of options for customizing the device. The dual Ethernet ports behind the phone support PoE for powering the device.The Cisco SPA502 Dubai phone can be expanded via available expansion modules. Or plug in a headset to the 2.5mm jack. The onboard speakerphone is perfect for local 3-way conferencing. Wideband audio offers high-quality voice and audio performance.
Call us today to know more about Cisco products and free consultation.We supply Cisco SPA502G in Dubai, Abu Dhabi, Sharjah, Fujairah, Ras Al Kaimah, Alman-UAE

 

Cisco SPA502G Specification

Supported network protocols:
  • MAC address (IEEE 802.3)
  • IPv4 (RFC 791)
  • Address Resolution Protocol (ARP)
  • DNS: A record (RFC 1706), the SRV record (RFC 2782)
  • Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
  • Internet Control Message Protocol (ICMP) (RFC 792)
  • TCP (RFC 793)
  • User Datagram Protocol (UDP) (RFC 768)
  • Real-Time Transport Protocol (RTP) (RFC 1889, 1890)
  • Real-Time Control Protocol (RTCP) (RFC 1889)
  • Differentiated Services (DiffServ) (RFC 2475)
  • Type of Service (ToS) (RFC 791, 1349)
  • VLAN tagging 802.1p / Q: Layer 2 quality of service (QoS)
  • Simple Network Time Protocol (SNTP) (RFC 2030)
Specification of the voice:
  • 1 line
  • V.2 SIP voice protocol (RFC 3261, 3262, 3263, 3264)
  • SPCP with the Cisco Unified Communications 500 Series
  • Support SIP server reduntantnego
  • Support for SIP networks with NAT (including STUN server support)
  • Registration redoing the primary SIP server
  • SIPFrag (RFC 3420)
  • The ability to initiate encrypted connections via SRTP
  • Voice codecs:G.711 (A-law and μ-law)
    G.726 (16/24/32/40 kbps)
    G.729 A
    G.722
  • Dynamic payload support
  • The ability to specify the size of audio frames per packet
  • Dual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO)
  • Flexible dial plan support with interdigit timers
  • Support for call by IP address / URL
  • Generating sounds during a conversation
  • Jitter buffer: adaptive
  • Hiding lost frames
  • Generate artificial background noise generation (CNG)
  • The algorithm recognizes a caller activity (VAD) algorithm damping silence
  • Adjusting damping / profit
  • VMWI – Voicemail waiting indicator, through subscriptions, notification
  • Third-party call control (RFC 3725)
Provisioning and Administration:
  • Built Administration Server to manage the administrator and user
  • Management by phone keypad on the screen
  • The system provisioning via HTTPS, HTTP, TFTP
  • Asynchronous notification for updates notification (NOTIFY)
  • Nonintrusive in-service upgrades
  • Generating reports and registration of events
  • Statistics BYE broadcast in the news
  • And syslog server entries dubug: configurable per line
Environmental Specifications:
  • Power supply:
    Power of Ethernet 802.3af
    The power supply is optional and is purchased separately
    Compatible power supply models: Cisco PA100-NA, PA100-EU, PA100-UK, PA100-AU
    Specifications power supply on the output: +5 VDC at 2.0A
Dimensions214 x 212 x 44 mm
Weight0.9 kg
Operating temperature0 ° ~ 40 ° C
Storage temperature-20 ° ~ 70 ° C
Air humidity Operating5% to 95%
Air humidity during storage5% to 95%