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Suncomm Lanpone 201
Suncomm SIP S777
SunComm

Suncomm Lanphone 201


Approx $115USD with a order of 50+ units.
Approx $104USD with order of 200+ units.
Approx $98 with order of 1000+ units.

These Phones run on a Motorola PPC 860 multimedia processor and run on VxWorks unix
They have a Kendin KS8995M intergrated network managed switch
The DSP is an AC4830
The PSTN interface is an AMS AS2525B

FEATURE:

ISP / ITSP (Internet Telepony Service Provider)
Phone to PC, Phone to Phone
Easy interface to ADSL / Cable Modem or Leased line equipment

TECHNICAL SPECIFICATION:

  • Built-in two 10/100 BaseT Switch / Hub
  • One RJ-11 PSTN analog line interface
  • Provide IP call or PSTN call selection (or) to be a POT set when external power is failure
  • Support ITU-T H.323 V2/V3/V4 protocol
  • PPPoE (option)
  • Behind NAT router or IP sharing device
  • Support both Fixed IP and DHCP
  • Support H.235 security function
  • Support Fast Start and H.245 Tunneling
  • Provide H.450 services (Hold, Transfer, Forward)
  • Multiple E.164 (line number) supported
  • Automatically GateKeeper Discovery
  • Alternate Gatekeeper selection
  • Provide Peer-to-Peer Mode (Non-Gatekeeper) selection
  • Speaker phone and Handset operation
  • 10 sets last number redial
  • LCD Display: Time, Date, Caller ID, Call Duration
  • 10 function keys for Memory dial or Multi-Line function
  • 8 one touch function keys : Speaker, Redial, Mute,
  • Hold, Transfer, Call Forward, PSTN, Message
  • 5 LED display : PSTN, Message, Hold, Mute and Speaker
  • Enter IP phone and Gatekeeper IP address from Keypads Configure Dial path selection
  • MS-NetMeeting v3.0 compatible
  • Support QOS by setting TOS (Type of Service), paramaters of VoIP packet
  • Bad Frame Interpolation
  • Provide both inband and H.245 Outband DTMF generation/detection
  • TFTP/FTP download
  • Two easy ways for system configuration: LCD Front Panel & TELNETImage

Suncomm SIP S777


I bought mine direct, so I don't know what MSRP actually is. However, I have seen this phone relabeled around Taiwan for around $100USD. Prices outside of Taiwan might vary greatly.

FEATURES:
Following RFC-3261
Dynamic IP support ( DHCP and PPPoE)
Passing through NAT devices
Remote software upgrade capability (via ftp)
Advanced Digital Signal Processing (DSP) technology to ensure superior audio quality
Support G.711 (A-law/U-law), G.723.1 (6.3k/5.3k), G.729A/B, voice codecs
Hand-Free Operation
Speed Dial: 10 speed dial keys
Volume Adjustment
Support supplementary services, including immediate(unconditional) call forwarding, busy call forwarding, no answer call forwarding and call transferring
Provide call history
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation)
Ping function supported
Call with or without proxy server (direct IP dialing)
Provide easy configuration methods
Support RFC-3261, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP, DNS, DHCP, NTP/SNTP, FTP, PPPoE protocols
Interoperable with most of the existing SIP VoIP devices ( IP-phone, gateway, proxy server)
Built-in switch function

SPECIFICATIONS:
PC Port: 1xRJ45 10/100 Base-T Ethernet, line auto sensing/switching
WAN Port: 1xRJ45 10/100 Base-T Ethernet, line auto sensing/switching
Power Over Ethernet 802.3af function (Option)
LCD display: 2x16 characters
Phone Case: 20-button keypad
Universal Switching Power Adaptor:
Input: 100-240V AC
Output: +12V DC, 350mA
Speaker: 8 Ohm/0.2 Watt speaker for speakerphone operation
Dimension: 17cm (W) x22cm (D) x 6cm (H)
Weight: 730g

Image


Similar to many voip-info members, I use Asterisk as my PBX. My configuration is super simple and standard:

sip.conf:

[78]
port = 5061           ; Port to bind to (SIP is 5060)
username=78
type=friend
secret=78
allow=ulaw
allow=alaw
allow=speex
insecure=yes
host=dynamic

extensions.conf:

exten => 78,1,Dial(SIP/78&SIP/,15)
exten => 78,2,Voicemail(78)
exten => 78,3,Hangup

[incomingTrunk]
exten => _++yellow:Your DID++,1,Answer
exten => _++yellow:Your DID++,2,Dial(SIP/78&SIP/15)
exten => _++yellow:Your DID++,3,Voicemail(78)
exten => _++yellow:Your DID++,4,Hangup


Phone configuration scripts (as per the manual):

Menu/OK -> Configure
Pass=135

Now you can use the UP or DOWN to set the needed configuration.

It is quite simple, so I won't go through all the steps here. But if you want the manual, send me an email.
Suncomm Lanpone 201
Suncomm SIP S777
SunComm

Suncomm Lanphone 201


Approx $115USD with a order of 50+ units.
Approx $104USD with order of 200+ units.
Approx $98 with order of 1000+ units.

These Phones run on a Motorola PPC 860 multimedia processor and run on VxWorks unix
They have a Kendin KS8995M intergrated network managed switch
The DSP is an AC4830
The PSTN interface is an AMS AS2525B

FEATURE:

ISP / ITSP (Internet Telepony Service Provider)
Phone to PC, Phone to Phone
Easy interface to ADSL / Cable Modem or Leased line equipment

TECHNICAL SPECIFICATION:

  • Built-in two 10/100 BaseT Switch / Hub
  • One RJ-11 PSTN analog line interface
  • Provide IP call or PSTN call selection (or) to be a POT set when external power is failure
  • Support ITU-T H.323 V2/V3/V4 protocol
  • PPPoE (option)
  • Behind NAT router or IP sharing device
  • Support both Fixed IP and DHCP
  • Support H.235 security function
  • Support Fast Start and H.245 Tunneling
  • Provide H.450 services (Hold, Transfer, Forward)
  • Multiple E.164 (line number) supported
  • Automatically GateKeeper Discovery
  • Alternate Gatekeeper selection
  • Provide Peer-to-Peer Mode (Non-Gatekeeper) selection
  • Speaker phone and Handset operation
  • 10 sets last number redial
  • LCD Display: Time, Date, Caller ID, Call Duration
  • 10 function keys for Memory dial or Multi-Line function
  • 8 one touch function keys : Speaker, Redial, Mute,
  • Hold, Transfer, Call Forward, PSTN, Message
  • 5 LED display : PSTN, Message, Hold, Mute and Speaker
  • Enter IP phone and Gatekeeper IP address from Keypads Configure Dial path selection
  • MS-NetMeeting v3.0 compatible
  • Support QOS by setting TOS (Type of Service), paramaters of VoIP packet
  • Bad Frame Interpolation
  • Provide both inband and H.245 Outband DTMF generation/detection
  • TFTP/FTP download
  • Two easy ways for system configuration: LCD Front Panel & TELNETImage

Suncomm SIP S777


I bought mine direct, so I don't know what MSRP actually is. However, I have seen this phone relabeled around Taiwan for around $100USD. Prices outside of Taiwan might vary greatly.

FEATURES:
Following RFC-3261
Dynamic IP support ( DHCP and PPPoE)
Passing through NAT devices
Remote software upgrade capability (via ftp)
Advanced Digital Signal Processing (DSP) technology to ensure superior audio quality
Support G.711 (A-law/U-law), G.723.1 (6.3k/5.3k), G.729A/B, voice codecs
Hand-Free Operation
Speed Dial: 10 speed dial keys
Volume Adjustment
Support supplementary services, including immediate(unconditional) call forwarding, busy call forwarding, no answer call forwarding and call transferring
Provide call history
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation)
Ping function supported
Call with or without proxy server (direct IP dialing)
Provide easy configuration methods
Support RFC-3261, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP, DNS, DHCP, NTP/SNTP, FTP, PPPoE protocols
Interoperable with most of the existing SIP VoIP devices ( IP-phone, gateway, proxy server)
Built-in switch function

SPECIFICATIONS:
PC Port: 1xRJ45 10/100 Base-T Ethernet, line auto sensing/switching
WAN Port: 1xRJ45 10/100 Base-T Ethernet, line auto sensing/switching
Power Over Ethernet 802.3af function (Option)
LCD display: 2x16 characters
Phone Case: 20-button keypad
Universal Switching Power Adaptor:
Input: 100-240V AC
Output: +12V DC, 350mA
Speaker: 8 Ohm/0.2 Watt speaker for speakerphone operation
Dimension: 17cm (W) x22cm (D) x 6cm (H)
Weight: 730g

Image


Similar to many voip-info members, I use Asterisk as my PBX. My configuration is super simple and standard:

sip.conf:

[78]
port = 5061           ; Port to bind to (SIP is 5060)
username=78
type=friend
secret=78
allow=ulaw
allow=alaw
allow=speex
insecure=yes
host=dynamic

extensions.conf:

exten => 78,1,Dial(SIP/78&SIP/,15)
exten => 78,2,Voicemail(78)
exten => 78,3,Hangup

[incomingTrunk]
exten => _++yellow:Your DID++,1,Answer
exten => _++yellow:Your DID++,2,Dial(SIP/78&SIP/15)
exten => _++yellow:Your DID++,3,Voicemail(78)
exten => _++yellow:Your DID++,4,Hangup


Phone configuration scripts (as per the manual):

Menu/OK -> Configure
Pass=135

Now you can use the UP or DOWN to set the needed configuration.

It is quite simple, so I won't go through all the steps here. But if you want the manual, send me an email.
Created by: hegars, Last modification: Thu 04 of Nov, 2010 (04:41 UTC) by admin
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