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.

FaramPhone

bodgey on 18 September, 2008 - 13:00
Keywords: Softphone

FaramTech

www.faramtech.com

www.faramtech.com

FaramPhone

Windows XP, Windows Vista, Windows 2000

Faram Phone is a SIP softphone which following the NGN, 3G network standards and our intention is to make it be the potential communication terminal in both NGN & 3G networks. User can use the softphone to make Audio, Audio & Video call and conference among these networks.

Sangoma Technologies - NetBorder Express Gateway Cards

stygmah on 17 September, 2008 - 17:40
Keywords: Gateway

Sangoma Technologies

www.sangoma.com

Sangoma Technologies - NetBorder Express Gateway Cards

Windows

Introducing the industry's first affordable TDM to SIP Gateway on a card.
Sangoma's NetBorder Express portfolio is a complete SIP-compliant VoIPi Media Gateway solution, delivering cost effective solutions for developers, system integrators, and OEMs in a convenient PCI or PCIExpress form-factor.


Mercuro IMS Client

bossiel on 17 September, 2008 - 13:29
Keywords: SIP | Softphone

Inexbee

Mercuro IMS Client

Windows 2000/XP/Vista

The Mercuro IMS Client is fully compliant with the main 3GPP IMS specifications and partially compliant with RCS (Rich Communication Suite) phase 1 and OMA specifications.

NS - 2300 GSM / VOWNET / WIFI SIP Mobile Phone

fajardofa on 18 July, 2008 - 14:27
Keywords: Miscellaneous

Network Sourcing Telecommunications Co., Inc.

NS - 2300 GSM / VOWNET / WIFI SIP Mobile Phone

Windows Mobile 5 (Upgradable WM 6.1 to WM7)

The first GSMi / VOIP WIFI Mobile in the world. It's the first mobile wherein the GSM and VOIP via WIFI is working on parallel. It's basically a complete laptop in a mobile. NS Telecom, a manufacturing company in the Philippines, has developed a work on-parallel GSM / VOWNET (Voice Over Wireless Network) mobile device, which is known to be the first in the world. With free NS to NS, NS to Gtalk, US, Canada, Singapore, Hongkong, UK call anywhere in the world!

NS - 3000 GSM / VOWNET / WIFI SIP Mobile Phone

fajardofa on 18 July, 2008 - 14:22
Keywords: Hardphone

Network Sourcing Telecommunications Co., Inc.

NS - 3000 GSM / VOWNET / WIFI SIP Mobile Phone

Windows Mobile 6.1 Touch Screen / Qwerty

The first GSMi / VOIP WIFI Mobile in the world. It's the first mobile wherein the GSM and VOIP via WIFI is working on parallel. It's basically a complete laptop in a mobile. NS Telecom, a manufacturing company in the Philippines, has developed a work on-parallel GSM / VOWNET (Voice Over Wireless Network) mobile device, which is known to be the first in the world. With free NS to NS, NS to Gtalk, US, Canada, Singapore, Hongkong, UK call anywhere in the world!

Lalala

LoekieBoy on 23 June, 2008 - 09:13
Keywords: SIP

Lalala

http://www.hro.nl

http://www.hro.nl

MizuPhone -a new nextgen SIP client

istvan3 on 30 May, 2008 - 14:58
Keywords: Softphone

MizuTech

http://www.mizutech.hu

http://www.mizutech.hu

MizuPhone -a new nextgen SIP client

Windows (2000,2003,XP,Vista)

How to add banner by programming

park on 29 May, 2008 - 11:21

how to setup sems ( ivr + voice mail)

ngovu on 5 May, 2008 - 08:08
Keywords: sems.conf | ser.cfg | System

0755

0755 on 30 April, 2008 - 15:41

12313

Step 1: Do something111


Gempro

www.gempro.com.tw

www.gempro.com.tw

GP-510 Fixed Bluetooth Mobile Terminal For 3G/3.5G/CDMA/WCDMA Bluetooth mobile

GP-510 Fixed Bluetooth Mobile TerminalModel number: GP-510GP-510 uses Bluetooth to connect Bluetooth mobile phone with common telephone or PBXi.It makes immediately PBX has a trunki by mobile phone to transmit.It is not only far away the electromagnetic wave, but also unnecessary to change user’s habit of using the table phone. It is a trunk of mobile to mobile, and it also saves expenses.Feature: 

  1. PBX/phone  Bluetooth Mobile  Voice is transmitted on the trunk by mobile phone.
  2. GP-510 pairs to Bluetooth mobile phone to use directly, it doesn’t relate to telecommunication provider and frequency.
  3. Just use phone to dial, it can be far away the electromagnetic wave.
  4. Speed Dial Function.
  5. The calling will pass to PSTNi automatically when power down, mobile phone is unpaired or out of range.
  6. Volume of digitization and it adjusts volume and gains.
  7. Set local code and automatically add code, and user isn’t necessary to change habit of using the table phone.
  8. Bluetooth has functions of pairing and searching automatically.
  9. Transmit high quality voice.
  10. It is compatible with GSMi/CDMA/3G/3.5G Bluetooth mobile phone.
Specification:Metering Signals: 12KHz, 16KHz,Polarity Reversal,ToneA/D ,D/A : G.711i:PCMi ModeDTMF : Detect & GeneratorCaller ID Format: DTMFTone Programmable: Dial Tone & Busy TonePower : AC110/220V 50/60HZBluetooth:Bluetooth Specification V2. Carrier Frequency 2400MHz to 2483.5MHz( USA , Spain ,France) Modulation Method GFSK,1Mbps,0.5BT GaussianOutput level, class 2 10 meters working range

GP-710 Bluetooth VoIP Gateway

gempro on 12 March, 2008 - 06:54
Keywords: Gateway | SIP

Gempro Technology Inc.

http://www.gempro.com.tw

GP-710 Bluetooth VoIP Gateway

[GP-710] is a revolution and innovation product.
It Integrated VoIPi and 2.4G Bluetooth technical in a device,
this not only offered high cost-effectiveness in mobile communication for mobile user,
but also offered very flexibility in re-building system.

Feature:
1. Could use with GSMi/CDMA/WCDMA/3G/3.5G
    various Bluetooth mobile phone.
2. VoIP and Bluetooth Mobile fully integration.
3. Compatible with SIP RFC543, RFC3261.
4. Support UPLINK , DOWNLINK Routining.
5. Support Voice report and Setting IP function
6. Offered Web Site for inquire or setting.
7. Support SIP PROXY, or point to point     application.
8. Offered one stage ,two stage free dialing     and call transfer function.
9. Bluetooth with auto pairing and auto     searching function.
10.QoS and Digital Transmit.

Specification:
VoIP
Web Browser
IVRi Interface
Uplink Route Setting
Downlink Setting
1 Stage, Dialout Called, Free Dial

eval

janakj on 11 February, 2008 - 16:22
Keywords: 2.0.x | eval | Man page

Standard

Eval module =========== Author: tomas.mandys at iptel dot org The module implements expression evaluation in route script, i.e. enables e.g. addition, concatanation, value items etc. There are two basic types: integer and string. Operation are processed using values on stack and polish notation. Besides the stack there are also register that may be accessed by quick manner in run time, via select or stack manipulation functions, because they are fixed during fixup phase. This depends on libuuid shared library. Module parameters: ----------------- declare_register: string; Declares one register, multiple declaration supported Example: modparam("eval", "declare_register", "ax"); modparam("eval", "declare_register", "bx"); modparam("eval", "declare_register", "cx"); modparam("eval", "declare_register", "dx"); xlbuf_size: int; Default: 4096 Size of buffer for xlib formating.

SEMS 0.10.0 voicemail server howto

Stefan on 28 January, 2008 - 20:25

HOWTO set up a voicemail server with SEMSi 0.10.0.


General

This document describes how to set up a voicemail server using SEMSi 0.10.0 release.
From previous versions the installation process has been simplified a lot for SEMS 0.10.0.
If you want to use SEMS 0.10.0 rc2 or SEMS 0.10.0 rc1, please have a look at the tutorial for rc2 or tutorial for rc1.

We will be using SERi 0.9.6-semsi as the SIP stack for SEMS. The SER 0.9.6-sems comes bundled with the SEMS 0.10.0 release, which makes installation pretty easy. Installation of SER 0.9.6-sems will be in a separate directory (/opt/seri-sems/) such that it can be living on the same host together with different SER versions.

VoxiPlus - VoIP-PRI-GSM Gateway

sorin on 28 January, 2008 - 13:20
Keywords: Gateway

Topex

http://www.topex.ro

Topex VoxiPlus is a media gateway. Its main function is to bridge between VoIPi networks ISDNi networks and Mobile networks. With Topex VoxiPlus you start saving from day one after installation by reducing the cost of communications.

VoiBridge - VoIP-GSM Gateway

sorin on 28 January, 2008 - 13:19
Keywords: Gateway

Topex

http://www.topex.ro

Topex VoiBridge is a VoIPi to GSMi/UMTS small capacity gateway. Its main functionality is to interconnect VoIP networks with mobile networks. With Topex VoiBridge you make significant savings on calls from IP to GSM networks and backwards.

tele

lbriceno6 on 19 December, 2007 - 10:45
Keywords: Softphone

luis

auth_identity

gergo on 10 December, 2007 - 14:37
Keywords: auth_identity | Module | 2.1.x

Standard

This module contains functions that are used for Enhancements for Authenticated Identity Management in SIP (defined by RFC4474). The purpose of this mechanism is securely identifying originators of SIP requests and providing integrity protection of the message, especially in an interdomain context.

Xeepe: SIP and ISDN Telephone System

silicatvalley on 27 November, 2007 - 12:14
silicatvalley's picture
Keywords: Gateway | PBX | SIP

Xeepe project

http://xeepe.com

Xeepe: SIP and ISDN Telephone System

Windows

Xeepe presents a fully integrated SIP and ISDNi VoIPi Telephone System

Xeepe has been developed especially for the needs of SOHO, small and medium sized companies, law & tax offices and Agencies. It is designed to work within the office LAN behind NATi and provides NAT traversal solution. Go test the fully functional trial version!


dp@xeepe.com

Business SIP Phone with PoE (802.3af), supplied from China manufacturer

unitatech on 26 October, 2007 - 07:41
Keywords: Gateway | Hardphone | PBX | SIP

Uni-Ta Technology Co., Ltd.

http://www.uni-ta.com.cn

Business SIP Phone with PoE (802.3af), supplied from China manufacturer

Model UTP-200: Business SIP Phone with PoE(802.3af); Support up to 5 SIP domains; 3 softkeys for flexible function definition, as well as a 5-position navigation key; Earphone jack; Easy installation and administration through web interfaces or keypad; Firmware and configuration values upgrading through FTP/TFTP; Business design…for more information, please click here http://www.uni-ta.com.cn/view.asp?prono=58

Analog Adapter compatible with SIP/IAX2

unitatech on 26 October, 2007 - 07:36
Keywords: SIP

Uni-Ta Technology Co., Ltd.

http://www.uni-ta.com.cn

Analog Adapter compatible with SIP/IAX2

Model UTA-601: VoIPi Adapter with single FXS and PSTNi back up. Compliant with both SIP & IAX2 protocol; Selective PSTN/VoIP voice communication; Support T.38 fax transmission; Support 2 SIP domains registering at the same time; Router built in; Reliable quality due to utilizing of high performance chipset; Smart design; Ideal device for VoIP carriers…for more information, pl

SIP phone supplied directly from China Manufacturer

unitatech on 26 October, 2007 - 07:32
Keywords: SIP

Uni-Ta Technology Co., Ltd.

http://www.uni-ta.com.cn

Model UTP-110: VoIPi phone supports up to 3 SIP servers registering at the same time, in very attractive designs. Excellent full-duplex speakerphone; Crystal sound quality due to utilizing of state-of-the-art DSP technology; Compatible with G.7xx codec; support NATi Traversal, VPN; Easy installation and administration through web interfaces or keypad; Ideal equipment for both office and home residential application….for more information, please click here http://www.uni-ta.com.cn/view.asp?prono=57

SIP phone supplied directly from China Manufacturer

unitatech on 26 October, 2007 - 07:32
Keywords: SIP

Uni-Ta Technology Co., Ltd.

http://www.uni-ta.com.cn

SIP phone supplied directly from China Manufacturer

Model UTP-110: VoIPi phone supports up to 3 SIP servers registering at the same time, in very attractive designs. Excellent full-duplex speakerphone; Crystal sound quality due to utilizing of state-of-the-art DSP technology; Compatible with G.7xx codec; support NATi Traversal, VPN; Easy installation and administration through web interfaces or keypad; Ideal equipment for both office and home residential application….for more information, please click here http://www.uni-ta.com.cn/view.asp?prono=57

SIP/IAX2 IP Phone

unitatech on 26 October, 2007 - 07:22
Keywords: SIP

Uni-Ta Technology Co., Ltd.

http://www.uni-ta.com.cn

SIP/IAX2 IP Phone

-Description: SIP/IAX2 IP Phone
-Model: UTP-102
-WAN: 1 X 10/100Mpbs RJ45 port
-LAN: 1 X 10/100Mpbs RJ45 port
-LCD display: 2 X 16 characters with back light
-Speaker: 8 Ohm/0.2 Watt speaker for speaker operation
-Standard: SIP RFC 3261, RFC 2543, IAX2
-Voice codec: G. 723.1 (5.3k/6.3k), G. 729A/B, G. 711(A-law/u-law)
-Voice standard:
Auto Gain Control (AGCi)
G. 168/165 compliant 16ms echo cancellation
Auto Echo Cancellation (AEC)
Voice Activity Detection (VAD)
Comfort Noise Generation (CNG)
In-band, out-of-band DTMFi relay, RFC2833, SIP info
Adaptive Jitter Buffer
-Call features:
Call forward, call transfer, 3-way conference call
Customized dial rules
Caller ID

WengoPhone

dneary on 2 October, 2007 - 15:00
Keywords: Softphone

OpenWengo

http://www.openwengo.org

Linux, Mac, Windows

Uni-Ta Technology Co., Ltd.

http://www.uni-ta.com.cn

Business SIP Phone with PoE (802.3af), supplied from China manufacturer

Model UTP-200: Business SIP Phone with PoE(802.3af); Support up to 5 SIP domains; 3 softkeys for flexible function definition, as well as a 5-position navigation key; Earphone jack; Easy installation and administration through web interfaces or keypad; Firmware and configuration values upgrading through FTP/TFTP; Business design…

SIP Developer Suite

Paul Glen on 25 September, 2007 - 13:53
Keywords: SIP

Radvision Ltd.

The award-winning SIP Developer Suite is a powerful and highly versatile set of tools to dramatically accelerate development of SIP applications. It includes a suite of Toolkits, Add-Ons and testing tools that enable developers to combine the necessary components for building an ideal development environment for an application's specific needs. The SIP Developer Suite complies with IETFi and 3GPPi standards, and is IMSi-compliant (3GPP and TISPAN). The high performance tools provide multiple API layers for full user control and flexibility.

The modular SIP Developer Suite is made up of "mix and match" of components. This enables developers to create the exact environment needed for specific applications, while retaining a small footprint and boosting performance. High Level APIs hide IMS and SIP complexity to accelerate development time.

Sippy Softswitch

sobomax on 21 August, 2007 - 21:43

Sippy Software, Inc.

http://www.sippysoft.com

Sippy Softswitch

The Sippy Softswitch is a carrier-grade Class 5 SIP session controller and rating/billing solution (residential&wholesale). It can act as SBC providing complete solution to the NATi traversal problem.  The product also provides extremely flexible customer management platform that enables the providers of IP Telephony services to launch, price, and provision an array of residential and wholesale VoIPi services.

IP phone, ATA, USB phone, WIFI voip phone

farseegroup on 21 August, 2007 - 03:30
Keywords: Hardphone

Farsee Technology Co., Ltd.

IP phone, ATA, USB phone, WIFI voip phone

IP phone supporting SIP, IAX2 protocols with FXO interface.

OEM is welcome.

Latest WIFI voip phone for choice.

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