Was ist "SipToSis" Skype Gateway?

SipToSis (Integrations-Software für Skype und SIP) ist eine Java-Software die es ermöglicht von Skype anrufe auf einem SIP Telefon entgegennehmen zu können, oder ein Anruf von einem SIP Telefon zu einem Skype Teilnehmer zu tätigen, z.B. über eine Asterisk PBX. Dies ermöglicht eine Integration von SIP zu Skype und umgekehrt. Im Grunde ist es ein Skype/SIP-Bridge/Gateway/Proxy/Adapter/-Konverter. Es ist also ein Codec-Konverter um SIP RTP-Audio in PCM-Audio kompatibel Skype und Skype-PCM-Audio umzuwandeln in SIP RTP-Audio. Damit wird eine Skype-to-SIP Signalisierung ermöglicht und führt SIP für Skype Anrufe im Umgang mit einem SIP-Adapter und einer Asterisk PBX durch.

Quellenlink: http://www.mhspot.com

SipToSis SIP to Skype Gateway Features:

    * Call Skype users using mappings/speed dial or use SkypeOut to make PSTN calls
    * Make SIP calls from Skype using a SIP provider or SIP PBX
    * Skype callers can be directed to the SIP address of your choice
    * SIP callers can be directed to the Skype user of your choice
    * SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks
    * Skype to SIP authentication/denial mappings via incoming Skype User ID
    * SIP DTMF touchtone decoding/encoding via RFC2833, INFO or Inband
    * Skype DTMF touchtone decoding/encoding via Inband
    * Connect Asterisk, FreePBX, Elastix, trixbox, PBX-in-a-Flash, 3CX or other SIP PBX to Skype Users
    * Conference call as of version 20091115 - See FAQ page for compatibility
    * Callback capability as of version 20091115 - See FAQ page for compatibility
    * Skype voicemail retrieval via SIP as of version 20091206
    * Outgoing Skype voicemail support as of version 20091206 - See FAQ page for compatibility
    * Auto play pre-recorded file(s) to SIP and Skype callers
    * SIP caller pin authentication and dialing
    * Skype caller pin authentication and dialing
    * PCMU (u-law)/PCMA (a-law)/G.711/iLBC included codecs and GSM with additional libs
    * Codec interface so you can add other codecs
    * SIP and Skype Hold
    * SkypeOut dialing rules - customize for your location
    * SIP outbound dialing rules - customize for your location
    * STUN support for public IP discovery
    * Call Time limiting ability
    * Usage limiting abilities by used time and unique called number count
    * Can be setup as a multi channel Skype to Asterisk Trunk for multiple simultaneous calls with the stsTrunkBuilder
    * Multiplatform (Windows/Linux/Mac OS X)
    * Run everything on a single computer if you wish
    * Windows users can run it as a service - see Appicus Windows Service Wrapper

SipToSis SIP to Skype Gateway System Requirements:

    * Skype client - See FAQ page for known working versions
    * Java 1.5 or higher (Linux users, use SUN's java)
    * SIP/VOIP adapter such as a SPA3102, SIP IP Phone, register with a SIP provider or setup an Asterisk/SIP PBX server
    * Skype4Java compatible platform
    * Sufficient bandwidth to support your configuration - Broadband preferred.

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