Sangoma S205 IP Phone in Addis Ababa
The Sangoma S205 is one SIP-Account VoIP with Dual Ethernet connectivity Support. It supports 5-way Conferencing without the use of Conference Bridges. Sangoma S205 Designed to work with FreePBX and PBXact. These Phones can use along with other popular IP PBX Platforms as well. Sangoma IP phones are so smart you can quickly and easily use them right out of the box.
Sangoma S205 Supported one SIP-Account and built highly economical price point. These VoIP Phones offer true Zero Touch Configuration by automatically locate FreePBX / PBXact and quickly get the full configuration right out of the box. Unlike other IP PBX systems, the redirection server automatically points the phone to the Sangoma FreePBX / PBXact for configuration.You don’t have to program the S205 with server Details. Instead, Zero Touch provisioning will do the Job for you.It will eliminate the need for manually configuring many different parameters on each phone and saves a lot of time and complexity.You can control global settings, program their phone keys, map extensions, upload images, download new firmware, and much more from the FreePBX platform itself.
The Hotdesking Feature in Sangoma S205 allows the users to switch locations or work from anywhere without hassle.The Users Just need to log in to any phone with your extension and password, and all your settings follow you instantly.
Sangoma S205 offered with Built-in VPN Client allows the users to securely connect to the telephone system from anywhere in the world.It is perfect for Perfect for remote workers to be able to access all their tools from the office without compromising security.
Sangoma S205 Specifications
- 1x SIP Account
- Dual 10/100 Mbps Ethernet Ports
- Full Duplex Speaker Phone
- Built-in VPN
- Power over Ethernet (IEEE 802.3af) (PoE), class 3
- Call forward, call waiting, call transfer
- 128 x 40 pixel graphical LCD
- 5-way Conferencing
- four context-sensitive “soft” keys
- HD voice, HD handset, HD speaker
- Illuminated LEDs for line status information
- LED for call and message waiting for indication
- Protocol SIP v1, v2
- Voice Activity Detection
- Auto Gain Control Comfort Noise Generation
- Acoustic Echo Cancellation
- Packet Loss Concealment
- Adaptive Jitter Buffer
- Full-duplex hands-free speakerphone with AEC (Acoustic Echo Cancellation)
- 1x RJ9 (4P4C) handset port
- Codec – iLBC, G.722, G.723, G.711(A/μ), GSM_FR, G.729AB, G.726 -32
- Auto-provision via HTTP / HTTPS, FTP / TFTP
- QoS – 802.1p/Q tagging (VLAN), Layer 3 ToS DSCP
- TLS (Transport Layer Security)