Cisco ip phone

Cisco Linksys SPA922 Single Line VoIP IP Phone - Imported Features: Dual-switched Ethernet ports, speakerphone, caller ID, call hold, and conferencing. 802.3af PoE, high-resolution graphical display, speakerphone, and a 2.5 mm headset port. Support for one line with two call appearances, as well as three-way conferencing, attended call transfer, and call hold. Easy station moves and shared line appearances across local and geographically dispersed locations. Standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Detailed performance measurement and troubleshooting. For more detail Call us 03455181377 03454797559

The Cisco SPA504g is a 4 line VoIP phone which features a large backlit LCD, 2 switched 10/100 ports, and Cisco HD Voice. Supporting either SIP or SPCP, the Cisco SPA504G is the perfect IP phone for hosted VoIP or premise based IP PBX systems. Cisco SPA504G Features & Functions: 4-Line Phone Dual switched ethernet ports IEEE 802.3af PoE (Power Over Ethernet) Supports SIP or SPCP protocol G.729 and G.722 Wideband Audio Support Menu driven user interface Web based GUI, remote provisioning IP dialing support Hi Res LCD Display 2.5mm stereo headset connector AUX port (SPA932 or SPA500S attendant console is supported) High quality audio handset Full duplex hands free speakerphone Hearing aid compatability (HAC) Support calls between TTY's Works well with GN Netcom, Jabra and Plantronics wired headsets Optional PA100 Power Adapter and MB100 Wall Mount Adapter

Digital IP Phone VoIP Phone Cisco 7961 Multi Line Quick Specs: Six line phone High-resolution 4-bit grayscale display (320x222) High quality, duplex speakerphone Integrated 10/100 switch for direct network connection Adjustable Foot stand 24 selectable ring tones Voice Activated Detection (VAD) HAC Compliant Pre-standard and IEEE 803.af PoE compliant For more detail Call us 03455181377 03454797559

Description: Details iShopping.pk brings you the best price for D-Link SIP Color LCD IP Phone (DPH-150GE/F5) with express shipping all over Pakistan.   Details: Internet phone made for home & office The DPH-120SE/F2 Broadband Internet IP Phone allows you to take advantage of your DSL/cable modem connection to make inexpensive Internet phone calls. It combines the industry’s latest Voice over the IP network (VoIP) technology with advanced communication features, and is compatible with industry- wide phone services. This sleek designed IP phone gives you the advantages of VoIP, while retaining the same look and feel of a traditional desktop telephone.   Superior voice quality The DPH-120SE/F2 IP phone incorporates Quality of Service (QoS) to ensure that voice received through the Internet is the same as or even surpasses that received on the ordinary phone. It supports many useful functions such as call transfer, caller ID display, 3-way conference and speed dialing to make it convenient to dial out or answer phone calls.   Surf internet while making internet phone calls The DPH-120SE/F2 comes with two Ethernet ports, one to connect to a DSL or cable modem, or to your broadband router. Once connected and logged onto the Internet, you are ready to make and receive calls. In addition, this IP phone provides another Ethernet port for connection to your PC, so you can talk on the IP telephone while browsing web sites, send e-mail, transfer les and do other network tasks on your PC.   Quality of service and security The DPH-120SE/F2 supports voice VLAN to isolate your voice communication so it cannot be tapped over the network. It provides priority queues to streamline trac so web browsing and Internet download will not interfere with voice trac and degrade Internet phone quality.

Description: Details iShopping.pk brings you the best price for D-Link Broadband Internet IP Phone (DPH-120SE/F2A) with express shipping all over Pakistan.   The DPH-120SE is a feature-rich telephone that provides voice communications using the same data network as your computer. This device combines the benefits of advanced Voice over IP technology with advanced communications features.   Specifications: Interfaces WAN Port: 10/100 BASE-T RJ-45 with PoE support for Internet connection LAN Port: 10/100 BASE-T RJ-45 for PC Headset: RJ9 Handset: RJ9 LEDs Power Display Large dot matrix LCD display Display Resolution: 128×48 pixels Keyboard 4 programmable soft keys Navigation keys Headset Volume adjustment (-/+) Redial Speaker phone Dial phone numbers (including * and #) CPU Broadcom SDRAM 16 MB Flash Memory 4 MB Power Plug AC power plug Phone features Vxworks OS Support SIP2.0 (RFC3261) SIP support 2 SIP servers, and backup SIP proxy servers Supports SIP UDP / TCP/ TLS SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer / IP Call Audio Features HD voice: HD handset Codec: G.711A/u, G.7231 high/low, G.729AB, G.722, G.726-32 DTMF: In-band, RFC 2833 and SIP INFO Full duplex hands- free speaker phone with acoustic echo cancellation Voice Activity Detection (VAD) Comfort Noise Generation (CNG) Background noise suppression Adaptive jitter buffers Packet loss concealment SRTP mute Advanced Features Call forwarding (busy/no answer/always) Call transfer (unattended/attended/semi-attended) Call holding Call waiting 3 way call conference BLF list Join call Call pickup Call park Redial Call back Call completion Hot desking Auto redial Predial SIP messaging, MWI Flexible dial plan Barring function for outgoing calls Do not disturb Auto answer Caller ID display CLIR(rejects anonymous calls) CLIP(to make an anonymous call) Dial without registration Call logs with missed calls/incoming calls/outgoing calls. Each support 300 records Phonebook: 500 records Speed Dial (abbreviatedВ dialing) Hotline/Warm-line Call rejection Black List/ White List Intercom Password dial, DTMF hidden Action URL/ active URI Voice codec setting for each SIP line LDAP, XML phonebook Web dial Keypad lock, Emergency call Network/ Security Features Bridge mode and route PPPoE for xDSL 802.3af PoE VLANпј€Voice VLAN/Data VLANпј‰ 802.1X NAT (NAPT) NAT transverse:Support STUN client DHCP client on WAN/ DHCP server on LAN Main DNS and secondary DNS server DNS Relay SNTP Client Firewall and DMZ Web management access restricted VPN (L2TP) ping, Trace route, telnet Maintenance & Management WEB, telnet and LCD menu management Management with different account right Recover firmware and factory reset Upgrade firmware  via HTTP, HTTPS, FTP, TFTP DHCP option66 and custom option for auto provisioning Auto Provisioning to upgrade configuration file with AES encryption TR-069 and Syslog Dimensions 250 x 205 x 60 mm 191,8 x 155,7 x 179,2 mm Weight 0.84kg 0.84kg Power Supply DC 5 V/600 mA, power adapter Ethernet cable (PoE) 802.3af Power consumption IDLE: 2.5W IDLE: 1.2W to 1.9W Active: 2.8W Active: 2.4W to 3.5W Operation Temperature 0 to 40 °C Relative Humidity 10% to 65% non-condensing Certification FCC CCC CE RoHS

The Fanvil C58 IP Phone is highly effective designed for a mid-range phone that can be used as an integrated phone or a standalone. It is a business class SIP phone with an integrated dual Ethernet switch, G.722 codec support, supports SRTP, support BLF, and 4 DSS keys. POE is optional. Fanvil C58 Key Features Vxworks OS Support SIP2.0 (RFC3261) and correlative RFCs SIP support 2 SIP servers, and backup SIP proxy servers Supports SIP UDP / TCP/ TLS SIP support SIP domain, SIP authentication (none, basic, MD5) , DNS name of server, Peer to Peer / IP Call Large dot matrix LCD display and soft keys make user easier to use Soft keys programmable Function keys programmable (each key have 2-color led indication) Supports RJ-9 headset Multi languages displayed http://dreamsnw.com/fanvil-c58-ip-phone Avalaible at Dreams Network & Technology (Pvt) Ltd. Contact: LHR: 04235784892 KHI: 02134710763, 02134711677, 02134721416

Fanvil X3 Series IP Phone are industrial masterpiece offers superior user experience and simplicity for home and office users with elegant appearance and intelligent software. X3 is positioning to be not only a telephone siting on a user’s desktop for communication, but also a nice artwork in your living room or office. X3P Colourful IP Phone Features The finest artwork in IP /telephone industry Economical and intelligent product concepts Easy installation and configuration Smart and friendly user interface Multi-user accounts and profiles Secure and complete provisioning protocols High Interoperability – Compatible with major platforms: 3CX, Asterisk, Broadsoft, Elastix, Zycoo, etc. http://dreamsnw.com/x3-x3p Avalaible at Dreams Network & Technology (Pvt) Ltd. Contact: LHR: 04235784892 KHI: 02134710763, 02134711677, 02134721416

KEYPAD FEATURES 4 Soft keys for doing more functions 10 DSS keys 4 Navigator Keys for navigating in configuration Hold Call forward Three-party conference Phone Book MWI Headset Redial 12 numeric keys with star & pound key Mute Vol-/Vol+ PHONE FEATURES Multi-user (6 SIP accounts) Caller ID display Call History: 300 Calls Phone book (up to 500 contact) Remote phonebook (up to 4 xml phonebook) Day/Time display Call/Time display 9 Selective Ring tones 9 Speed dial number Incoming call indicator Flexible dial map Password control for Configuration + Pre-dial before sending Connect with expansion module MWI SMS Keypad lock Emergency call Customize DSS key/ softkey NETWORK INTERFACE AND I/O PORTS WAN Port: to connect to 10/100/1000Mbps Ethernet LAN Port: to connect to 10/100/1000Mbps Ethernet 5V DC IN Jack: for switching power adaptor Handset Hook switch: for hang on/off control on handset cradle Handset Jack: to connect with a handset Headset Jack: to connect with a headset External console Jack: to connect with expansion module VOICE CODEC G.711a/u (64k bps) G.729A/B (8k bps) G.723.1 high/low G.726-32 G.722 ADVANCE VOICE QUALITY FEATURE Silence Suppression Acoustic Echo Cancellation (G.167) Voice Active Detection (VAD) Comfort Noise Generation Jitter Buffer DTMF Transmitter (SIP info, Transparent, RFC 2833) Packet Lost Concealment SIGNAL, MEDIA & NETWORK PROTOCOLS SIP RFC 3261 & the related RFC standard in Appendix A SDP RFC 2327, RTP RFC 1889 IP assignment: Static IP, DHCP and PPPoE STUN, static port mapping (for NAT traversal) SNTP, DNS & DNS SRV TFTP/FTP/HTTP/HTTPS for Auto Provision IP/TCP/UDP/ARP/ICMP Route and Bridge mode SUPPLEMENTARY CALL FEATURE Call Hold Resume, Mute, Waiting, Call waiting Indication Three Way Conference, Anonymous Call/Rejection Message Waiting Indication, Auto Answer, Black list, Limit list Auto hang-up, Auto Redial, Ban outgoing, Hotline, BLF/Presence Action url/Active uri NETWORK CAPABILITY QoS: IEEE 802.1Q & IEEE 802.1p Compliant Diffserv (DSCP)/ToS, Full range VLAN ID Support Class of Service Support by VLAN Tag, LLDP, L2TP VPN/ OpenVpn USER INTERFACE AND NETWORK MANAGEMENT LCD/Keypad UI in English & other Languages HTTP(WEB) UI in English version & other Languages FTP/TFTP/HTTP for Firmware remote update Auto-provisioning (APS) for firmware and profile upgrade

· Multi-user (4 SIP accounts) · Caller ID display · Call History: 300 Calls · Day/Time display · Call/Time display · 9 Selective Ring tones · Incoming call indicator

· Integrated PoE · 3-way conference · Integrated dual-band WiFi · 2.4” (320x240) TFT color LCD

· 4 Lines · HD Audio · Busy Lamp Field · Encryption security · Swappable Faceplate

· Caller ID display · 2.8 inch Color LCD · Acoustic Echo Cancellation · Multi-user (4 SIP accounts)

· 1 x 10/100Mbps PoE support · 1 x 10/100Mbps LAN port · VLAN support · 1 SIP Line · 1 Line LCD display

Cisco 7962 Double Line Digital IP VoIP Phone Brand: Cisco Model: 7962 Features: Connectivity Type: IP/VoIP Line Appearances: 6 Speakerphone: full duplex Network Connections: 2x 10/100 Ethernet PoE: yes Protocol Support: SIP, SCCP Display: 5" monochrome, 320x222 Audio: wideband

Cisco 7912 Unified Voip Phone Large Pixel-based display - gives access to features. 4 dynamic soft keys - Changeable calling options for the user to scroll through. Menu key - This key allows users to quickly access information such as call logs and phone settings. Hold key - visual indicatior that a call was placed on hold. Support of a single line or directory number Support of Extension Mobility Call monitor (speaker only, no microphone) Messages key - access to voicemail messages Four speed dials configurable at the Cisco Unified CallManager Cisco Discovery Protocol Differentiated Services Code Point (DSCP) tagging G.711a, G.711u, and G.729ab audiocompression coders-decoders (codecs) Integrated Ethernet switch 10/100BASE-T Ethernet connection through an RJ-45 interface for LAN connectivity Software upgrade supported using a Trivial File Transfer Protocol (TFTP) server Provisioning of network parameters through Dynamic Host Configuration Protocol (DHCP) Voice activity detection, silence suppression, comfort-noise generation, and error concealment Compatible with Cisco Unified CallManager Version 3.3(2) and later, using the Skinny Client Control Protocol (SCCP) protocol as well as SIP (RFC 2543)