Grandi IP phone, ATAs and VoIP gateway

easy3call on 8 November, 2006 - 14:36
easy3call's picture
Keywords: Gateway | H.323 | Hardphone | SIP

Grandi Digital Information Ltd.

www.easy3call.com

Grandi IP phone, ATAs and VoIP gateway

ARM 9

The Grandi GIP300 IP phone is a new entry level IP phone. GIP300 has dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NATi router/DHCP server, high quality full duplex hands-free speakerphone with acoustic echo cancellation, headset jack, message waiting indicator, and more memory for future function growth. It supports G.711i, G.723.1i, G.729A/B. GIP300 is able customer to register two SIP servers simultaneously. In addition, you could download music ringing tones and the GIP300 provides as many as eight kinds of ring tongs.
 
The Grandi GIX100 VoIPi ATAs  works great. It is very easy to configure, excellent audio and great selection of codecs including G.711, G.723.1, G.729A/B. It is inexpensive with dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NAT router/DHCP server and support message waiting indicator. And most importantly, GIX100 is an ultra-affordable VoIP ATAs with high performance.
 
Both GIP300 and GIX100  is a new generation VOIP terminal products which were developed based on single ARM9E CPU platform. Its core is a ARM9E CPU at 150MHZ. With this technology, Grandi would provide the products with high performance and ultra-affordability cost.
 
GIP300 is just at $30.00 USD for 100K units order.
GIX100 is just at $28.00 USD for 100K units order.


levent

iistr.net

iistr.net

GreenJ

ligraz on 18 November, 2011 - 08:29
Keywords: Softphone

Lorem Ipsum Medien Ges.m.b.H.

http://loremipsum.at

http://greenj.org

Platform independent

GreenJ is an open source Voice-over-IP phone software using pjsip and Qt. It can easily be used to build your own VoIPi phone system. Our approach was not to build a complete phone with user interface, but instead provide an application that handles only the communication. The program logic and user interface are separated from the application by using an integrated browser. We use webkit as browser engine, which is well integrated into Qt (QWebView). A Javascript interface handles all communications between application and webpage.

test

xnbnl0 on 8 November, 2011 - 13:43
Keywords: SIP

test

test.com

test

voxalot fher

axxessotel on 26 October, 2011 - 00:01
Keywords: SIP

us.voxalot.com

www.voxalot.com

www.voxalot.com

voip

Ipkall

Blissboi on 18 September, 2011 - 22:06
Keywords: Miscellaneous | Softphone

Ipkall.com

Www.ipkall.com

Www.ipkall.com

Phone1

Blissboi on 18 September, 2011 - 21:44
Keywords: Miscellaneous | Softphone

Ipkall

Www.ipkall.com

Www.ipkall.com

karl

karl on 23 August, 2011 - 02:44
Keywords: SIP

iptel.org

http://www.iptel.org

http://www.iptel.org

satoshun

hhqffx on 18 July, 2011 - 07:20
Keywords: SIP

satoshun

google

google

WeSoFly Entertainment Phone

wesoflyent on 18 July, 2011 - 03:34
Keywords: SIP

WeSoFlyEntertainment.com

OPAL

Robert Jongbloed on 20 May, 2011 - 07:15
Keywords: Protocol Stack

OPAL

http://www.opalvoip.org

Multiple

Open source H.323 and SIP stacks.

liantong

weizx2000 on 14 May, 2011 - 11:05
Keywords: Miscellaneous

liantong

www.chinaunion.com

www.chinaunion.com

iptel

zzw on 7 April, 2011 - 07:24
Keywords: SIP

iptel

iptel.org

iptel.org

Frank

frankieboy on 25 March, 2011 - 06:53
Keywords: SIP

Frankieboy

www.mastering.fi

www.mastering.fi

sip

hurri on 11 March, 2011 - 00:48
Keywords: SIP

hurri

iptel.org

iptel.org

IP VoIP Phonex

rouldph56 on 3 March, 2011 - 02:19
Keywords: Miscellaneous

StilTechCo LLC.

We deal with offering Business and Residential unlimited long distance world wide VoIPi!

lhl7998

lhl7998 on 2 December, 2010 - 13:59
Keywords: SIP

lhl7998

sip.iptel.org

sip.iptel.org

IMSDroid

Regicidal1 on 26 November, 2010 - 01:20
Keywords: Miscellaneous | SIP

Idk

Idk.com

idk.com

Signal server

fireman on 18 November, 2010 - 16:39
Keywords: Signaling Server

Polycom

Polycom.com

109.68.45.6

thepit

wifiducky on 17 November, 2010 - 06:30
Keywords: Testing

me

www.youtube.com

www.youtube.com

Sonetel free, global phone system

Henrik Thome on 14 September, 2010 - 08:53
Keywords: PBX

Sonetel

http://sonetel.com

http://sonetel.com

Windows/Mac/Linux/Android/Iphone

The Sonetel phone system is a free hosted PBXi service for unlimited users/extensions that can be connected to any third party phone companies and phones.

The free plug'n play Windows softphone (Sonetel Client) has chat, presence etc. and a management interface for advanced configuration of the Phone System.

Optional services from Sonetel include phone numbers (from +50 countries) and low cost calling to any landlines/mobile phones.

Service activation is instant and you get an international phone number for free testing.

VOIP SIP SDK

itspecialist on 7 September, 2010 - 08:39
Keywords: SIP

VOIP SIP SDK

http://ipphonesdk.com

http://ipphonesdk.com

Win 2000/XP/Vista

 SIP Client SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/RTPi compliant soft phone with a fully-customizable user interface and brand name. 

The VoIPi SIP Client SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either and delivers superior voice quality by integrating digital voice processing features including auto gain controller (AGCi), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMFi, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more! 

VoIP SIP Client SDK is based on IETFi standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SERi, Sip EXpress, OpenSER and Asterisk.

  New features of the VoIP SIP Client SDK: 

• g729 and g723 Codeci´s support
• Multiple and single Codec selection support
• Failure codes support (get SIP Message Response Code, SIP Message Response Text)
• RTP/RTCPi Port setting (for inbound RTP traffic)
• Reduce audio latency and audio latency settings (properties: MinPrefetchCount,    MaxPrefetchCount, MaxRTPPackets)
• Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted,    OnRemoteMediaStoped)
• Get used codec per line
• Custom Ringtone (play wav) support (property: RingtoneFile)
• Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine)
• Redirect Call to other phone line
• Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration)
• Complete new, re-written and updated samples with source code
• and much more!

 Here is a list of the main features of the VoIP SIP Client SDK:


• Easily make and receive SIP (Session Initiation Protocol) based phone calls through any    SIP gateway or SIP compliant IP-Telephony service provider
• VoIP conferencing with crystal clear sound even for both low and high-bandwidth users 
   G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBCi, L16 and g729 & g723 Codec
• Open standards-based and interoperable with all of the major equipment vendors
• UDP and TCP support for sip activex
• Multi-party voice conference support/ Conference split and join, locally mixed conferences
• Multi-line support (multiple simultaneous calls)
• SIP Instant/Chat Messaging with send/receive controlling with direct sip activex support
• Integrated STUN, TURN and ICE support
<• Comes with new sample SIP Proxy Server    to provide in bundle with the SIP    Client ActiveX a ready up own SIP VoIP    and Instant Messaging network solution.
• P2P support for directly connections    between 2 SIP clients( sip activex ) without SIP Server
• Outbound proxy server support
• Encrypted SIP account settings (encrypted    SIP account settings in your webpage)
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices
   on-the-fly - also during a conversation/    conference)
• Mute microphone/speaker + level indicator
• Auto-answer
• DNDi (Do Not Disturb)
• Adaptive Jitter buffer
• PLC (Packet Lost Concealment)
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or    suppression)
• Noise cancellation or suppression
• DTMF tones support (generation/detection)
• Recording voice conversation into PCMi    WAVE (.wav) file
• Playing PCM WAVE (.wav) files to the    remote end
• Audio file memory cache
• Extended SIP URL functions
• Dynamically loadable codec support (coming soon)
• Comes as ActiveX control (Web demo with ready-up signed CAB included)
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Microphone and Speaker Volume with Mute support
• Keep-alive packets to NATi/firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Keep-alive packets to NAT/firewall
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate 
• Fully sample applications for various programming languages such as sample source code    for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and Delphi
• For .NET framework as well and all development environments with ActiveX support

Easy, familiar, event-driven call control ActiveX
• Easy to use; quick development 
• Support for .NET framework and all development environments with ActiveX support
• Very easy to incorporate

Rich call control feature set
• Multi-party voice conference support (Conference split/ join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• SIP Instant messaging 
• Locally mixed conferences
• Hold/Mute 
• Call transfer 
• Call forwarding and rejection

Industry leading SIP support
• RFC3261 compliant SIP stack 
• RFC 2833 out-of-band DTMF signaling
• Integrated STUN, TURN and ICE support

Comprehensive configuration support
• Select media input/output devices (on-the-fly as well during a conversation/conference)
• Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE) 
• SIP proxy

Advanced digital voice processing features
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression

… and much more!

Llamadas Gratis

roderickzapata on 21 July, 2010 - 01:14
Keywords: SIP

Doddle Phone

www.paypallatino.com

.NET SIP Library

JamesWright on 15 July, 2010 - 05:58
Keywords: Protocol Stack | SIP

Konnetic

www.konnetic.com

.NET/Mono

A set of RFC-compliant high-functionality class libraries which manage SIP messaging. Designed to support the development of cross-platform multimedia and text-based applications and services written any any .NET language (e.g. c#, VB or F#).

Linphone

TJipson75 on 15 June, 2010 - 17:02
Keywords: SIP

Simon Morlat

http://simon.morlat.org

www.linphone.com

Android

7404

gnsn on 26 April, 2010 - 15:14
Keywords: H.323

Bilion

7404

IP/TEL

najmid1 on 19 April, 2010 - 20:39
Keywords: SIP

LOCPAZ

http://192.168.1.33

http://192.168.1.33

1921.168.1.33

//

najmid@yahoo.fr

Wuchuan Network (Shenzhen) Limited

www.5111soft.com

Single port with PSTNi VoIPi gateway supporting SIP/IAX2 protocol, support VPN, auto provisioning, VLAN. Via enterprise VPN and IP private line,it supports simultaneous access of IP audio and date service.Besides,it is gateway with PSTN suitable for VOIP service provider, can work with Asterisk IP PBXi perfectly.

Support Protocol:

Support SIP (RFC3261, RFC2543)
Support IAX2
Support Reverse polarity
Support voice code: G711A/u, G729, G726, iLBCi
Support G.168 echo cancellation standard, compliant 96ms echo cancellation with speaker
Support Jitter Buffer, VAD, CNG, SIP/IAX2, domain name register, point to point communication.
Support the voice communication through RTPi and RTCPi.
Support the DTMFi Inbound/Outbound transmission; SIP info, DTMF Relay, RFC2833
Support standards of the ring in different countries and regions.
NATi penetration, support STUN client, CITRON, AVS etc..
Support SIP domain, SIP Authentication (none basic, MD5), domain name parse.
Support two SIP servers (Public Server / Private Server) synchronously, make a call by either proxy.
Support SIP application, including SIP Call forward/transfer/holding/waiting.
Support L2TP protocol

Network Features:

Support two models: Bridge and Router, and integrates the router functions of these two ports.
Support basic NAT and NAPTi
Support PPPoE for xDSL, and support redial automatically when its offline
Support DHCP client for WAN port
Support DHCP server for LAN port
Support DNS relay for LAN port, and provide DNS service for LAN network devices.
Use the advanced technology DSP to ensure high quality voice
Use the advanced buffer technology to avoid the information package delay too long or lost.
Support Network Tools that includes ping, trace route and telnet client
Support 3 methods to configure the WAN IP: Static (static configuration for LAN), DHCP (Dynamic query through LAN) and PPPoE (Dynamic query through ADSLi)
Provide firewall control for small size of LAN port
Provide the communication PRI available for the small LAN of LAN port
Support Qos (802.1p) for the second Layer

Advanced Functions:

It can register to two SIP server synchronously,one IAX2 systems and a PSTN number, it has 3 VOIP number and 1 PSTN number, that means one phone has 4 numbers
Call waiting, call transfer, 3 ways calling, many modes call forward
Switch outgoing call freely between VOIP and PSTN
Incoming display, forbid call out; setup avoid-disturb, auto answer, off hook auto dial, quick dial
Setup blacklist number and limit number
Support point-to-point call
Setup the end number method
Setup fixed call method
Support Silence Suppression, VAD (Voice Activity Detection)
Support CNG (Comfort Noise Generation)
Support Echo Cancellation and AGCi (Automatic Gain Control)
Support Polarity reversal
Support VPN (L2TP), can be used in the VPN special network
Support SIP register failure detection, offer busy tone, when the telephone is out of order
Support line detection, busy tone when no line available

Configuration, management and maintenance:

Support post mode and can update gateway via post mode
Support to operate and configure the gateway by keyboard and Http mode, and support the filtration restriction on user IP address
Support to update the software and configuration via HTTP, FTP, TFTP
Support multi-administrator management, user name, password; Support reversal Telnet through NAT/ Firewall to manage long-distance.


Wuchuan Network (Shenzhen) Limited

www.5111soft.com

A newly developed hi-tech SIP phone for enterprise supporting 5 SIP servers synchronously,(POE) function and 128*64 LCD display.Besides, it supports 3 interactive soft key to play humanized operating prompt and an option of a pair of headsets for private calls.


Support Protocol

Support SIP (RFC3261, RFC3262, RFC2543).
Support Voice codec: G711A/u, G729, and G723.1
Support G.168 echo cancellation standard, compliant 96ms with speaker mode.
Support Jitter Buffer, VAD, CNG, SIP Domain name register, point-to-point Call
Support RTPi and RTCPi
Support the Inbound/Outbound transmission; SIP info, DTMFi Relay, RFC2833
Support many countries' standard ring
NATi transversal: Support STUN, CITRON, AVS Mode
Support SIP domain, SIP Authentication (none, basic, MD5), Domain Name parse
Support 5 SIP servers synchronously, can call in and out by either proxy
Support SIP application, including SIP call forward/transfer/holding/waiting

Network Features

Support two models: Bridge and Router, integrate two ports router function.
Support basic NAT and NAPTi.
Support PPPoE for xDSL, and support off hook auto dial.
Support DHCP Client for WAN;
Support DHCP server for LAN;
Support DNS relay for LAN and can provide DNS service for LAN Network equipment.
Use advanced DSP tech to insure high quality voice
Use advanced jitter buffer tech to prevent the delaying and losing for package information
Support Network Tools, including ping, race route, and telnet client.
Support three modes to configure WAN port IP, they are: static, DHCP, and PPPoE.
Provide firewall control for small LAN.
Provide optional communication priority level for small LAN.
Support Secondly Layer QoS(802.1p)
Support 12tp

Advanced Function

Support headset
Support 128*64 LCD
Support Power over Ethernet (POE) function
Support 5 SIP servers synchronously.
3 Interactive soft key to play humanized operating prompt.
Support local voice record, message and server message record.
Support message wait indication.
Support user defined ring tone.
Support L2TP client.
Support call pickup, join call, auto-redial.
Support 5 programmable keys, 5 PSTNi keys and 5 SIP keys, and it can be connected with the expansion board which can display more numbers' online status.
Support dial switchboard and extension number at one time to get through extension directly.
Support call list, and can set different rings according to different incoming callers.
Dial waiting, call transfer, three ways call, and multi-dial forward
Caller ID display, ban calling out, setting no-disturb, dial number automatically while picking up the telephone, set VIP numbers.
Set the black name list and confine numbers
Support point-point calling directly.
Support flexible methods of receiving numbers.
Support silence suppression and silence detection.
Support noise background simulation.
Support echoes suppression and auto gain. 


Wuchuan Network (Shenzhen) Limited

www.5111soft.com

An award-winning next generation IP phone supporting SIP/IAX2 and online status display.Featuring for VPN,VLAN,superb sound quality and rich functionalities at ultra-affordable price,it is a special phone for IP PBXi,can display presence status.

Support Protocol

SIP (RFC3261, RFC2543)
Support IAX2.
Support codec: G.711A/u, G.723 high/low, G.729i A/B
Support G.168 echo cancellation standard, compliant 96ms echo cancellation with speaker mode
Support voice volume adjustment, including IN/OUT of handset and speaker
Support Jitter Buffer, VAD, CNG, SIP, Domain name register, point-to-point Call
Support RTPi and RTCPi
Support the Inbound/Outbound transmission;SIP info,DTMFi Relay,RFC2833
Support many countries' standard ring
Support NATi: Support STUN, CITRON, AVS Mode
Support SIP domain, SIP Authentication (none, basic, MD5), Domain Name parse
Support two SIP server synchronously, including Pubic Server/ Private server,can make a call by any proxy. You can back-up and select any above SIP server.
Support SIP application, including SIP Call forward/transfer/holding/conference/pickup/redial/unredial/joincall
Support BLF, server presence and Peer to peer presence negotiation
Support VPN (L2TP) clients

Network Features:

Support two models: Bridge and Router, integrate two ports router function.
Support basic NAT and NAPTi
Support PPPoE for xDSL, and support auto redial when disconnect
Support DHCP Client for WAN
Support DHCP server for LAN
Support DNS relay for LAN and provide DNS service for LAN Network equipment
Support DNS domain name resolution in WAN port
Support SNTP Client to get time from internet
Support advanced DSP tech to ensure high quality voice
Support advanced jitter buffer tech to prevent the info package delaying and losing
Support network tool: ping,trace route,telnet client
Support three modes to configure WAN port IP, they are: static, DHCP, and PPPoE
Provide firewall for small LAN
Provide optional priority level for small LAN
Support second layer QoS (802.1p)
Support VLAN
Support VPN, L2TP protocol. (New hardware support openVPN)

Special and Advanced Function:


With IP PBX,support 10 group quick dial number, together with IP PBX presence subscribe, the IP phone can display directly the online state of the booking numbers by the indicator. If the indicator is green, means it is online; the indicator is green and twinkle, means it is in the course of the call; if red, means it is offline.
Support the local voice message, play the message by one key, IVRi personality record the message, voice prompt.
Caller ID display, ban calling out, avoid-disturb setting, auto-answering, auto dial while picking up the telephone, quick dial;
Call waiting, call transfer, three ways call, and multi-dial forward
Setting the black list and limit numbers
Support point to point call
Setting the ended number methods
Setting the ended number add, delete and substitution
Setting the fixed calling ways
Support phone number
Support Silence Suppression, VAD (Voice Activity Detection)
Support CNG (Comfort Noise Generation)
Support Echo Suppression and AGCi (Automatic Gain Control)
Support DIGEST validate and MD5/MD5-sess encapsulation
Support local/server Message-Waiting Indication
Support auto long-distance configuration and edition auto-upgrade.
Support call record checking and management.
Support call pickup
Support join Call
Support redial and unredial
Support directly dial IP+port to call SIP terminal device