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Skype Gateways

Created by: jht2,Last modification on Wed 01 of Jul, 2009 [18:35 UTC] by GregD
A list of products that can be used to connect non-PC devices to the Skype network.
Skype itself is available for Windows, Mac OSX, Linux and Pocket-PCs, but for many of the products are at this time only windows drivers available.

Page Contents


Opensource and non-commercial

Most Gateways are commercial. Please move opensource tools here

Overview

  • chan_celliax and skypiax
  • SipToSis
  • MyToGo for Skype


chan_celliax

-Celliax (free, GPL) is a channel driver for the Asterisk Free PBX that manages GSM and CDMA cellular phones through an adapter, composed by a datacable (for commands) and an audiocable (for the voice) interfacing the computer soundcard.
-chan_celliax is also capable of making and receiving Skype calls through the cellphone, and has an app like app_directory that let you choose which one of your Skype contacts you want to call.
-Booting in console mode you find a script (/usr/local/asterisk/usr/sbin/celliax_server) that automatically launch Xvfb (the fake X11 server), Skype and Celliax. Also via ssh!
-Together with chan_celliax is distributed the Celliax LiveCD, with a working installation of Asterisk, chan_celliax, and configuration utilities based on Knoppix. The Celliax LiveCD contains also all is needed to run Celliax on Windows:the cygwin installer

skypiax

See here for FreeSwitch and Asterisk
Read more about implemenation tips for Windows in this thread.

MyToGo for Skype is free

See http://saveonphoneservices.com/modules.php?name=News&file=article&sid=9

What is it? and What can it Do?

Currently, for example, Skype To Go, is only available in the following countries:

Australia, Chile, Denmark, Estoina, Finland, Ireland, new Zeland, Poland, Sweden, United Kingdom, USA.

Because MyToGo for Skype uses your Online Number ("Formally called SkypeIn") MyToGo Works in these additional countries, as well:

Brazil, Dominican Republic, France, Germany, Hong Kong, Hungary, Italy, Japan, Korea Republic of, Mexico, Romaina, Switzerland.

Additionally, Skype To Go calls to these countries are currently blocked. Algeria, Belarus, China, Egypt, Kazakhstan, Kuwait, Lebanon, Macedonia, Morocco, Nigeria, Romania, Russia, Ukraine, Vietnam, Serbia and Montenegro.

MyToGo does NOT block these countries.

Of course, MyToGo for Skype can be used in any country that has Online Numbers ("Formally SkypeIn") as well.


SipToSis: Skype To SIP integration software gateway/bridge/proxy/converter using Java (Linux, MacOS, Windows)

MHSpot LLC introduces SipToSis - SipToSis is an open source java application that allows you to make and receive Skype calls to and from SIP devices or networks.

  • Make and receive Skype calls from your SIP device/PBX
  • Make and receive SIP calls from Skype
  • Java Open source under GNU GPL license
  • Multiplatform
  • Supports SIP DTMF via RFC2833, INFO and Inband
  • Supports Skype DTMF using Inband
  • Authentication mappings and routing
  • Dialing Rules/Transforms
  • Incoming SIP Pin authentication and Skype destination dialing
  • Works with Asterisk/Tribox/Elastix/3CX and other SIP PBX's
  • Can be used as a Multi Channel Skype trunk
  • Call time limiting features
  • Daily call limitig features
  • Free download

Get the SipToSis Software SIP to Skype Gateway for free.

SipToSis (Sip to Skype integration software) is Java software that allows you to make and receive Skype calls from your SIP/VOIP telephone adapter, SIP IP phone or Asterisk/SIP PBX. You can also make and receive SIP calls from Skype. It lets you integrate Skype into your SIP VOIP phone system. Basically a Skype/SIP Bridge/Gateway/Proxy/Adapter/Converter. It has a codec converter to convert SIP RTP audio to compatible Skype PCM audio and Skype PCM audio to SIP RTP audio. It performs SIP signaling and Skype call handling to connect with your SIP adapter, Asterisk Server, SIP PBX or SIP VOIP provider.

Why wait? Build your own for Asterisk

Nerd Vittels has a HowTo for FreePBX and SipToSis


Commercial software



Overview

  • Skype for Asterisk (SFA) by Digium
  • FaxIP for Skype from NetGen Communications, Inc.
  • PrettyMay Call Center for Skype
  • Skip2PBX (Skype gateway)
  • Uplink (Windows only)
  • Skype Asterisk Channel: chan_skype (Linux)


Skype for Asterisk by Digium

Currently still in closed beta (as of April 2009) this will be the first native Skype gateway for Asterisk.
See here and here for more details.

Some of the features that will be supported in the market release are:
  1. SkypeIn: Receive calls from the public telephone network using standard phone numbers
  2. SkypeOut: Make calls to landline and mobile numbers at incredibly low rates
  3. Standard phone features: Incoming/outgoing digits (DTMF), Caller ID
  4. Smart call routing based on called Skype Name, Caller ID, country of the caller, language they have chosen in their Skype client and etc.
  5. Retrieve Skype credit balance information
  6. Store and call PSTN and Skype contacts
  7. Retrieve and set Skype user presence information
  8. Support for G.711 and G.729 voice codecs
  9. Each Skype channel license includes a Digium G.729 codec license


FaxIP for Skype from NetGen Communications, Inc.


FaxIP for Skype allows users to send real-time (machine-to-machine) faxes anywhere in the world for free. The initial release requires that the two parties have FaxIP installed. FaxIP is comprised of a hardware component that converts the fax terminal's RJ-11 to the PC's USB port. The FaxIP software interfaces with the converter on one side and Skype on the other. Future versions will include support for software-only configurations and faxing to non-Skype users.

Read more....

If you have interest, please contact us.

NetGen Communications, Inc.
1225 Northmeadow Pkwy
Roswell, GA 30076
770-449-7704 Phone
770-242-7050 Fax
sales@netgencommunications.com


PrettyMay Skype PBX Gateway


Skype Gateway to PBX solution from PrettyMay enables business of all sizes to make and receive Skype calls without breaking the existing phone systems, simply use it as a Skype trunking solution for your existing office PBX system, then save money on the cheap Skypeout calls.

PBX_Gateway-180.png


With Sangoma analog (FXS/FXO) cards supported, PrettyMay Skype PBX Gateway supports up to 24 simultaneous lines on one server PC with Windows Xp/Server 2003/Vista/Server 2008 installed.

With a FREE demo version of the software, and 30-day "no questions asked" return policy on the hardware, it is 100% risk free during your 30 days trial period.

A Skype Gateway to PBX solution offers numerous benefits:

  • Lower phone bills for employees and callers.
  • Rapid return on investment.
  • Works with any Analog/IP PBX.
  • Free branch office connection.



Learn more at: www.prettymay.net


Skip2PBX (Skype gateway)


Completely cross-platform; (Built with Linux)

Skip2PBX is a Software designed with self-contained, modified Linux core, sets up in independent PC and interfaces with Skype for all types (SkypeIN SkypeOUT Skype to Skype) of Skype Calls specialized for Business as a Skype to PBX gateway, interfacing through FXO, FXS, ISDN (BRI-PRI) and SIP.
Exclusive features not found in other produced gateways, including one that can utilize all 30 of the maximum lines in system using only one main Skype account number. This particular product supports Eicon Diva, Digium, and Sangoma cards. USB and SIP support also possible. Further Information: Skype to PBX Gateways

Read a
review of SIP-Skype gateways to PBX systems. Linux, Network Security, Skip2PBX

Read a
review of Skype-SIP software gateways: Uplink, PSGW and chan_skype

Stonevoice to launch Skystone (SIP & H.323)

Skype Gateway: Completely IP based (SIP & H323) and software (plug&play) with embedded evaluation. Skystone interconnects the skype world directly to the IP PBX or Legacy PBX (through a media gateway)


Uplink (Windows only)



PSGw: Skype to SIP and H.323 converter (Linux and Windows)

RSDevs.com presents PSGw - Personal Skype to SIP and H.323 gateway) PGw is an application that allows to connect Skype network with H.323 and SIP networks. PSGw works as router that should be placed between Skype and H.323 or SIP network. PSGw can route calls from Skype users to H.323 or SIP endpoint according to user-defined rules.
  • PSGw Info
  • Integrate Skype into your working enviroment, but only for windows. -> According to rsdevs.com sales: linux version in 2006
  • PSGw works with enterprise PBX like Alcatel.
  • PSGw allows to provide free support over Skype.
  • PSGw 3.5 for Windows including DTMF relay
  • PSGw 1.6 for Linux Info: So far without DTMF relay
  • Integrate Skype into your working enviroment with SIP servers
  • PSGw for Linux is compatible with Asterisk, however note that SIP port 5060 is hardcoded and cannot be changed
Tips: Look at Xvfb (fake X11 server), Xvnc (eliminates the need for X forwarding) and snd-dummy (dummy sound device)


Skype Asterisk Channel: chan_skype (Linux)

Enables Skype to be intergrated with Asterisk. http://www.chanskype.com
  • Be warned that the Skype client needs to run on the same box as Asterisk, but VNC works wonders; since X is not recommended on any serious Asterisk PBX due to timing issues you will probably need to install a separate Asterisk to act as SIP-Skype gateway
  • Multiple Skype instances can be run
  • DTMF relay works!
  • Commercial product
  • Click here for a HOWTO for installing ChanSkype on Trixbox


Commercial software gateways

Overview

PSTN to Skype Services

  • DID World Wide forwards international numbers from more then 50 countries to Skype.

SIP to Skype Services

  • SkypeTrunk Convergence Between SIP and Skype Services. Supports SIP/H.323 to Skype protocol conversion, 2 channels for free to test connection quality.

OpenSky

OpenSky is a free service provided by Gizmo5 which allows any mobile phone, web browser or IP aware phone network (SIP, ASTERISK, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls.

OpenSky is a Hosted solution, the quickest and easiest solution for all of your users to connect to any Skype user. To use OpenSky, there's:

  • No Hardware to buy
  • No Software to buy
  • No Software to install
  • No Skype accounts required
  • Free (1-5 minute calls)

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From Business/SIP/Asterisk Networks:


Any Asterisk or SIP network can use OpenSky by simply having their users dial Skype identities like this: SkypeName@opensky.gizmo5.com.
This will route the call though Gizmo5's OpenSky gateway. These are free calls for 1-5 minutes. To make longer calls, it is easy for system administrators to purchase OpenSky and enable it for all users.

Setting up for Asterisk Server:


[gizmo5]
type=peer ;COPY THIS CONFIG
host=198.65.166.131 ;INTO YOUR sip.conf
fromdomain=proxy01.sipphone.com ;THIS WILL
canreinvite=no ;ALLOW ANY
nat=yes ;DEVICE OR CLIENT
dtmfmode=rfc2833 ;CONNECTED TO YOUR
insecure=very ;ASTERISK SERVER TO CALL
qualify=yes ;SKYPE USERS SEVERAL WAYS.
fromuser=YOURSIP ;BY DIALING SKYPE NAMES OR NUMERIC SHORTCUTS
authuser=YOURSIP ;ENTERED INDIVIDUALLY BELOW
username=YOURSIP ;OR BY DIALING skype_skypeusername
secret=YOURPASS ;OR THE 333 ALIASES
disallow=all ;ENTERED at my.gizmo5.com
allow=ulaw ;
allow=alaw ;SEE gizmo5.com/opensky
allow=ilbc ;FOR MORE INFO

[general]
exten => _1333.,1,Goto(opensky,,1) ;COPY THIS CONFIG
exten => _333.,1,Goto(opensky,,1) ;INTO YOUR
exten => _skype[_].,1,Goto(opensky,,1) ;extensions.conf
exten => 563,1,Dial(SIP/skype_echo123@proxy01.sipphone.com) ;To dial a Skype user by dialing 563 in this example echo123
exten => echo,1,Dial(SIP/skype_echo123@proxy01.sipphone.com) ;To dial a Skype name in this example echo will dial echo123

[opensky]
exten => _1.,1,NoOp('opensky dial')
exten => _1.,2,Dial(SIP/${EXTEN}@gizmo5|120|j)
exten => _1.,3,Hangup()

You can also call from any browser, any pc, and any mobile phone



Commercial hardware


Overview

Cordless Phone Gateways

Connects a phone device, but has no FXS port.

Single Analog FXS/FXO Gateways

Connects a phone device and has FXS port too. One can switch between a FXS call and a skype call by pressing a * or #)
  • Atcom: au-600 1 fxs 1 fxo,forward skype to your mobile phone, make skypeout call with your mobile phone
  • Ipmate
    • S-90: 1 FXO, 1 FXS, 1 USB port. Recieve and make normal and skype calls on/from your current inhome telephone network. ( at the time Windows only)
    • GW-90: Route incoming skype calls (via USB) to a FXO or a FXS port. + Route incoming FXS calls to Skype (via USB) or FXO port. ( at the time Windows only)
  • ActionTec
  • Plextel USB Skype Gateway with 1FXS connect to phone/cordless phone/PBX/PABX and 1 FXO to telephone line, allow forward skype call to normal phone/PABX
  • Teleon Networks - 1 port PCI FXO card for Skype. Freeware driver to make common voice modem as a Skype gateway.

Enterprise level: Multi-channel skype gateways

  • PrettyMay Skype PBX Gateway supports Sangoma A200/B600 series cards with up to 24 FXS/FXO ports, works on Windows Xp/Server 2003/Vista/Server 2008 system.
  • IndustryDynamics. VoiceGear Connect Skype Gateways with up to 24 FXS/FXO/mixed analog ports, ISDN PRI and SIP connectivity. Complete integration of Skype into any corporate PBX. More info.
  • AirTouch AirTouch Communication Services delivers Enterprise Strength, High Density Skype Gateways with E1/T1 lines interface or FXO/FXS analog ports.
  • VoSKY Exchange Uses WinXP and up to 4 skype accounts to deliver 4 FXS ports to a PBX
  • MESSiX - Multiline Enterprise Software Skype Exchange from the writers of PSGw. Supports Skype to SIP/H.323/IAX2 protocol conversion in software for 4 channels and will be expanding to at least 120 channels
  • Teleon Networks - Skype Soho PBX TSP-S104D, 1 CO line, 4 extensions with 1 PCI Skype interface.
  • http://skip2pbx.com: Requires a dedicated machine (Linux) with telephony card (BRI, PRI or USB FXS/FXO)



IndustryDynamics VoiceGear Connect Standalone Skype Gateways

VoiceGear Connect is a revolutionary standalone Skype Gateway that allows for complete integration of existing PBX systems with Skype network. VoiceGear Connect combines parallel Skype calling into a highly scalable package targeted towards large organizations and enterprise users. These gateways support most existing PBX connectivity options including up 24 FXS/FXO/mixed analog ports, ISDN E1/T1, or SIP connectivity to Asterisk or any other SIP PBX.

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  • Skype Trunking
VoiceGear Connect allows business of all sizes to leverage Skype, the world's largest VoIP network, as a trunking solution for their existing office PBX systems. Utilizing Skype trunking as an alternative to the PSTN, businesses can substantially save on both domestic long-distance and international calls.

  • SIP and ISDN PRI Connectivity to the PBX
VoiceGear Connect offers SIP connectivity to most IP and hybrid PBX systems utilizing SIP G.711 codec. Native support for SIP connectivity extends the ability of VoiceGear Connect to work with Asterisk based PBX systems by delivering high quality voice for up to 60 Skype channels.

  • Extension-side (FXS)/Trunk-side (FXO) and Mixed Connectivity to the PBX
VoiceGear Connect offers a variety of connectivity options to any PBX system with available extension-side (FXS) or trunk-side (FXO) ports. We also support mixed connectivity option by utilizing both FXS and FXO ports available on the PBX. Mixed FXS/FXO connectivity is unique to VoiceGear Connect products and is essential to implementing advanced calling features. Dual FXS/FXO connectivity also simplifies installation and lowers your costs if all ports are in use by allowing the addition of line cards instead of more expensive trunk cards.

  • Multi-site PBX Communication and Toll-bypass
Employees in multiple branch offices equipped with VoiceGear Connect gateways can communicate with each other completely free using Skype. Calls between all extensions in branch offices equipped with VoiceGear Connect are carried over the Skype network without incurring any fees.

  • Global Call Transfers over Skype
All incoming landline and Skype calls can be transferred to any phone number or Skype ID, inside or outside the company office. Calls can be transferred at no cost to employees in branch offices equipped with VoiceGear Connect gateway, or to employee's mobile phones and landline numbers at low SkypeOut rates. (requires extension side connectivity to the PBX).

  • Web Click-To-Call
VoiceGear Connect brings you Click-To-Call feature, which allows customers to call your company PBX directly from your Website. Customers browsing your Website can reach Customer Service, Sales or Support departments absolutely free over the Skype network with a simple click of a button added to your Website.

Other Features...

  • Easily scalable to support up to 60 concurrent Skype calls per gateway
  • Complete integration of Skype into your existing office PBX system allowing for handling incoming/outgoing Skype calls via your existing phones
  • Works with existing office PBX system to provide other telephony features such as Call Transfer, Call Hold, Call Waiting, Call Pickup, Conference Call and more... (PBX dependent)
  • Completely free Skype-to-Skype calls over existing phones and compelling long-distance services with SkypeOut
  • Call recording in MP3 format
  • Skype conferencing
  • Web user interface for complete system administration and Phonebook management
  • Complete set of Web-based management tools including call log and line status monitoring
  • Enhanced security with a built-in firewall and central administration of all Skype services. No need to run Skype clients on individual office PCs
  • Intelligent call routing with a single corporate Skype hunt-group for automatic distribution of all incoming Skype calls
  • Skype contact speed-dials for easy dialing of Skype users from regular desk phones
  • User authentication via Web interface and DTMF tones
  • Role-based user settings and custom dial plans
  • Incoming Caller ID Generation
  • Automatically finds a free line/port for the incoming and outgoing Skype call
  • Call and line monitoring with busy tone generation and detection to prevent line locking
  • Automatically detects On-hook and Off-hook signals
  • Programmable voice prompts to allert callers
  • Excellent sound quality with hardware based echo cancellation and noise reduction
  • Hardware monitoring and configuration via Web interface
  • Call control via DTMF tones, allows to place calls on hold, resume, record, create conference
  • Complete call logging and reporting on per authenticated user basis
  • Regular system updates delivered via the Web

Visit IndustryDynamics for more info

SkyBOX Professional

19" Rackmount Call Router,Gateway,Bridge for Skype & PSTN


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1.Benefits from Skype Unlimited Call & PSTN Reliability

2.No needs for extra PC


SkyBOX

19" Rackmount Call Router,Gateway,Bridge for Skype & PSTN

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Benefits from Skype Unlimited Call & PSTN Reliability


USKY UBS-400 Skype Gateway


UBS-400 is a 8 analog(4*FXO+4*FXS)Skype gateway which is designed specially for small and medium business users.


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Main Benefits:
1. Users can call SkypeOut from table phone directly without any habits changed.
2. By using also Skype Unlimited Calls Service, users can call 10,000 minutes by paying only $5.95 per month.
3. Users can choose to PSTN route by pressing # buttong on the table phone.
4. Seamlessly integrated with company existing PBX. No need to buy extra trunk card.
5. Build up global call center for company immediately.
6. Build up virtual branch in almost 20 country by using SkypeIn function.

Please visit www.globalysky.com for more information.

Super Skype Gateway


The Best Solution to Take the advantages of Skype Unlimited Calls to Landlines


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Callfree Gateway 1600 Professional - CG1600PRO

  • One of the best LCR (Least-Cost-Routing) Trunks for your company
  • Much Cost-Effective than ALL other Skype Gateways
  • 16-FXS Ports in One Single Box
  • Auto-CDR (Call Detail Record) Generation for Billing System
  • 4U-high Rackmount
  • Unlimited World with only USD 12.95
  • Apply to 34-Countries Landlines (Skype.com)
  • Apply to 5-Countries Mobiles (Skype.com)


PORTech Skype GSM Gateway / Skype Gateway build in PC

PORTech Communications Inc. Is a manufacturer in Taiwan. PORTech major in GSM Gateway, GSM Fixed Wireless Terminals (1,4,8 ports), VOIP GSM Gateway(1,2 ports), E1 GSM Channel Bank, Skype GSM gateway and Remote power switch by PSTN and Web.
http://www.portech.com.tw

SkyTrunk ST-1004~4 ports Skype, FXO, FXS, GSM
  • Skype to PBX, PBX to Skype free,
  • Skype to GSM, GSM to Skype
  • Skype to PSTN,PSTN to Skype
  • GSM Frequency: Dual band: 900 / 1800MHZ, Tri band: 900 / 1800 / 1900MHZ

You can set ST-1004's route as follws:
Skype to FXO,Skype to FXS,Skype to GSM
FXO to skype,FXO to FXS
FXS to skype,FXS to FXO,FXS to GSM
GSM to Skype,GSM to FXS

When GSM or PSTN or Skype call in,you both can set 2 stage dialing or transfer specific number

Example:
When GSM in,you can choose
(1) PBX or Phone ring
(2)2-stage dialing: provide dial tone,you can enter skype id or any phone number(phone number) or PBX
(3) transfer to specific phone number via skype out

St-1004 preferences:
1. Use skype service combines with your existing pabx / pbx system that is already in place.
2. Use the pabx / pbx extension, so you can dial skype out and receive skype in.
3. Skype to pbx or pbx to skype is free service.
4. Corporate users may have multiple skype connections meaning you can have more users online.
5. One company with one dial in number, skypbx automatically switches from a busy line to a free line. Works like a traditional pbx, centrall office lines.
6. Our staff may use the pbx system to dial out using skype for free.
7. You can manage skype accounts into one system.
8. Skypbx can appoint an user to manage the entire system. As to prevent any network intrusion and avoid any possibility of security concerns.
9. One pc can run 4 skype accounts meaning that at any time on one pc you have 4 lines available.
10. Skypbx provides you with the use of skype, fxo, fxs, gsm and interface router.

Product specification:
1. Communication agreement: complies with skype.
2. Firewall / nat agreement: complies with skype.
3. Interface: 4 FXS connections to the pbx system, 4 FXO connections to the telco lines and 4 mobile trunk connections for gsm interface.
4. USB port: 2. 0 USB port to pc.
5. LED indicator: every line from the skypbx has its own LED display.
6. Voltage: itu standard of 48 volts.
7. Dialing tone: standard busy, ringing and dialing tones.
8. Dtmf: auto detection and auto generate.
9. Managemetn tools: console setup of management software.
10. Input: dc12v, 2a.
11. Case size: 1u 228 (d) *437. 6 (w) *44 (h) mm.

Skype Gateway-SkyTrunk: Skype to PBX free, PBX to Skype free

SkyTrunk ST-1004~4 ports skype,FXO,FXS,GSM
GSM Frequency :Dual band:900/1800MHZ,Tri band:900/1800/1900MHZ
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SkyTrunk ST-2004 ~4 ports Skype,FXS
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SkyTrunk ST-3004~4 ports skype,FXO,FXS
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SkyServer:~4 ports Skype,FXS (Build in pc)
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If you have interest,please contact us

PORTech Communications Inc.
150 Shiang Shung N.Rd,Taichung,Taiwan
http://www.portech.com.tw
E-mail:info@portech.com.tw

Spintronics in voip-info.org

Scalable Skype PBX Gateway for HUGE Enterprise, up to 120 FXS ports or higher

Grid Controller
  • Scalable System: Use SkyGrid System to upgrade the Capacity of Skype PBX Gateway and save the investment
  • Unified Console for Call management & Administrator Interface
    • Phonebook/Prefix-handle for Inbound call
    • Group Hunting for Outbound call
    • Real-time setups all Configurations for all Skype PBX Gateway
  • Network Management: Monitor the status by Web-Administration
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Callfree Gateway
    • Concurrent Multiple Skype account working properly
    • Cluster Technology
    • No needs for extra PC
  • Connect to your company PBX and take advantages of Skype/SkypeIn/Skypeout services
  • Employee can use Skypeout for low-rate call
  • Customers can use regular phone and call SkypeIn number for local call service
  • Manager can use NB/PDA+Skype to call back to Company for free
  • Skyper can call customer service for free
  • Use Skype account: spin_callcenter ext.21 to feel it and ask the details
  • People who live in USA can use regular phone to call +1-415-508-3685 ext.21 to feel it and ask the details
  • People who live in Europe can use regular phone to call +44-20-7193-5503 ext.21 to feel it and ask the details
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Bring the full Connectivity to Callfree Gateway on any SIP/H.323 IP phones/IP PBX
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E1 PRI Skype PBX Gateway for Telecom Trunk Standard Interface E1 PRI
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All Spintronics products link

Skype compatible Softphones

  • www.jajah.com New all-in-one softphone which supports SIP, IAX, Skype (free) and POTS (low rates). Calling Skype users is possible without having Skype installed, but being called via Skype requries a Skype client running in parallel.


Skype to GSM Gateways

Lets you talk for free or at very low rates throughout the world from your mobile phone
  • MobiGater Make Skype/SkypeOut calls from your mobile phone. Receive Skype/SkypeIn calls on your mobile phone


PORTech Communications Inc. Is a manufacturer in Taiwan. PORTech major in GSM Gateway,GSM Fixed Wireless Terminals (1,4,8 ports), VOIP GSM Gateway(1,2 ports), E1 GSM Channel Bank, Skype GSM gateway and Remote power switch by PSTN and Web.
http://www.portech.com.tw

Skype GSM Gateway : SkyTrunk ST-1004~4 ports Skype, FXO, FXS, GSM

  • Skype to PBX, PBX to Skype free,
  • Skype to GSM, GSM to Skype
  • Skype to PSTN,PSTN to Skype
  • GSM Frequency: Dual band: 900 / 1800MHZ, Tri band: 900 / 1800 / 1900MHZ

You can set ST-1004's route as follws:
Skype to FXO,Skype to FXS,Skype to GSM
FXO to skype,FXO to FXS
FXS to skype,FXS to FXO,FXS to GSM
GSM to Skype,GSM to FXS

When GSM or PSTN or Skype call in,you both can set 2 stage dialing or transfer specific number

Example:
When GSM in,you can choose
(1) PBX or Phone ring
(2)2-stage dialing: provide dial tone,you can enter skype id or any phone number(phone number) or PBX
(3) transfer to specific phone number via skype out

St-1004 preferences:
1. Use skype service combines with your existing pabx / pbx system that is already in place.
2. Use the pabx / pbx extension, so you can dial skype out and receive skype in.
3. Skype to pbx or pbx to skype is free service.
4. Corporate users may have multiple skype connections meaning you can have more users online.
5. One company with one dial in number, skypbx automatically switches from a busy line to a free line. Works like a traditional pbx, centrall office lines.
6. Our staff may use the pbx system to dial out using skype for free.
7. You can manage skype accounts into one system.
8. Skypbx can appoint an user to manage the entire system. As to prevent any network intrusion and avoid any possibility of security concerns.
9. One pc can run 4 skype accounts meaning that at any time on one pc you have 4 lines available.
10. Skypbx provides you with the use of skype, fxo, fxs, gsm and interface router.

Product specification:
1. Communication agreement: complies with skype.
2. Firewall / nat agreement: complies with skype.
3. Interface: 4 FXS connections to the pbx system, 4 FXO connections to the telco lines and 4 mobile trunk connections for gsm interface.
4. USB port: 2. 0 USB port to pc.
5. LED indicator: every line from the skypbx has its own LED display.
6. Voltage: itu standard of 48 volts.
7. Dialing tone: standard busy, ringing and dialing tones.
8. Dtmf: auto detection and auto generate.
9. Managemetn tools: console setup of management software.

10. Input: dc12v, 2a.
11. Case size: 1u 228 (d) *437. 6 (w) *44 (h) mm.

Skype Gateway-SkyTrunk: Skype to PBX free, PBX to Skype free

SkyTrunk ST-1004~4 ports skype,FXO,FXS,GSM
GSM Frequency :Dual band:900/1800MHZ,Tri band:900/1800/1900MHZ
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SkyTrunk ST-2004 ~4 ports Skype,FXS
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SkyTrunk ST-3004~4 ports skype,FXO,FXS
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SkyServer:~4 ports Skype,FXS (Build in pc)
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If you have interest,please contact us

PORTech Communications Inc.
150 Shiang Shung N.Rd,Taichung,Taiwan
http://www.portech.com.tw
E-mail:info@portech.com.tw


See also



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by spamblock, Monday 13 of April, 2009 [20:18:03 UTC]
There is a nice little page that lists all removed advertising, called "Removed Advertising". Would be an idea to check that for reasons for a ban. Rules are quite simple, behave. You pushed your product BizPBX against Guidelines, even after request to stop reposting. You can whine about removals afterwards, but you really should have thought about his earlier.
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333Re: Why spamblock always removed Yeastar's announce here?

by yeastar, Thursday 12 of March, 2009 [07:11:56 UTC]
Spamblock, you are always aiming at Yeastar. You said we always reposted among others. Ok, I moved our announces on the end of page behind others. You said our news takes with 'New Releases', but you failed to tell us for which product we describled it as a new releases. And it's an imposition to removed Yeastar's all other announces like WiKi paga and homepages just because of you thought thye are 'Spams'. Excessively you left improper words 'Don't consider buying from them!!’ to misadvise others. Please (kind) stop your actions of removing, removing and more removing.
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333Re: Why spamblock always removed Yeastar's announce here?

by spamblock, Saturday 07 of March, 2009 [11:32:58 UTC]
Yeastar, read the comments for a change and think twice when a posting is removed instead of reposting, reposting and more reposting. As said, your company is now banned and any future postings will be removed.
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333Re: Why spamblock always removed Yeastar's announce here?

by yeastar, Wednesday 04 of March, 2009 [00:59:04 UTC]
Spamblock, it's your first time that (kindly) tell us the reason why you always against Yeastar. But after I carefully went over my whole content, I failed to find out any 'new' word. Now I am going to upload the announce again.
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333Re: Why spamblock always removed Yeastar's announce here?

by spamblock, Monday 02 of March, 2009 [07:24:18 UTC]
Yeastar has repeatably violated Guidelines by actions like re-posting 'new releases' among others. (Kind) requests to adjust this behavior has had no effect.
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333Why spamblock always removed Yeastar's announce here?

by yeastar, Tuesday 13 of January, 2009 [01:56:19 UTC]
Why spamblock always removed Yeastar's announce?

Spamblock, I'm a webmaster from Yeastar.com. And I announced a real news yesterday on Skype Gateways, we really have obeyed the rules after ADBLOCK modified the page.

PLEASE TELL ME OBVIOUS REASON FOR REMOVING OUR SKYPE GATEWAYS.

Looking forward for your answer!

Jessica

Jessica@yeastar.com
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333

by andr2, Wednesday 30 of January, 2008 [11:41:44 UTC]
Please, advise how end-users should make calls from Skype to some international phone number at PSTN.
Would be the best to get examples for 2 cases below:
1. Skype user with nickname "test1234" wants to call international PSTN-number +9714321234.
2. The person at PSTN with international number +9714321234 wants to call the Skype-user with nickname "test1234".

What number should be typed to make call to Skype and what numbers will be preseneted in both cases in Skype and for PSTN user.
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333ChanSkype

by DKvoip, Thursday 23 of November, 2006 [11:18:55 UTC]
ChanSkype works just perfect on UBUNTU.
Just follow very easy install instructions.
-
Missing DTMF functions.
Missing Version for * 1.4.x

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333Asterisk+Skype

by timhunt, Tuesday 31 of October, 2006 [13:45:33 UTC]
I tried

- Uplink - effectively requires a dedicated PC for each Skype channel. I couldn't maintain a stable configuration on a general purpose PC

- PSgw - SIP port is hardcoded at 5060 so requires a complicated Network configuration, if you plan to run multiple instances, or run it on an Asterisk server.

- ChanSkype - Not perfect but the only one that I was able to get working reliably.
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333IPMate W90

by idynamics, Tuesday 17 of October, 2006 [20:49:28 UTC]
IndustryDynamics is a Canadian-based reseller of IPMate W90 device. The device lets you forward PSTN calls to Skype and vice versa. For more info, visit us at: www.industrydynamics.ca