Skype Gateway

This gateway can be used for phone calls from WebClient WebPhone to Skype users.

Configuration of this gateway is based on the SipToSis project: http://www.mhspot.com/sts/siptosis.html

It is a Java Skype SIP gateway and is for free. It supports running multiple instances and multiple Skype instances too. A Skype client is required to be installed on the machine.
It can be easily integrated via VoIP gateways and users can just use both their soft-phones or hard-phone as usually.

How to Set Gateway

There is an old free version of SipToSis from December 2009. (The new version is paid – $2.50.)

Closely follow the guidelines how to setup SipToSis in the ATA Setup section. Basically, you just install Skype, extract SipToSis and run it via a batch file on Windows. Then you need to make a few configuration changes:

  1. In the siptosis.cfg file:

    Set the port where SipToSis should be bound.

    e.g. to port 5080:

    ...

    host_port=5080

    ...
    Do not change anything else.

  2. In the SkypeOutDialingRules.props file:

    This file specifies the rules to dial out. You do not need to touch it. It already contains something and there are simple rules you can skip and use our VoIP gateway rules instead.
    The SipToSis acts as a SIP client which when called will direct a call to the alias of the SIP URI (e. g. echo123@127.0.0.1:5080).

  3. In the SkypeToSipAuth.props file:
    This is a file for specification of incoming Skype calls and how they should be handled.

    It can be set e. g. to:
    *,sip:siptosis@127.0.0.1

    Which means forward calls to sip:siptosis@127.0.0.1

    The IP address of 127.0.0.1 (without a port) will contact the local SIP server and the address will be matched to the VoIP Gateway Contact header and will be linked with the Skype gateway setting on your VoIP server.

  4. Setting up an IceWarp VoIP Skype gateway:

    Create a new gateway:

    Server: 127.0.0.1:5080 (required)

    Proxy:

    User: siptosis

    Pass: siptosis

    Outgoing rules:

    Condition: ^(1)(.*)@(.*)$

    Rewrite: $2@127.0.0.1:5080

     

    This means that if a number starts with "1", take everything after than and call via the SipToSis gateway.

    e.g.: 1echo123 would call echo123 on Skype.

  5. Restart SipToSis, start Skype and you can make and receive your first calls.