Where's the documentation or manual? There isn't any "documentation" or manual. Setup can be simple or complicated depending on your capabilities and what you are trying to accomplish. There are many comments in the configuration files to help with settings. I have created some setup guides to help with basic setup:

• For SIP phone/SIP Adapters/ATA or single channel Asterisk/PBX SipToSis Phone Adapter Setup.
• For multi channels with a PBX Asterisk Skype Trunking Setup Guide
• For specific issues see SipToSis Configuration Troubleshooting.

What is the stsTrunkBuilder and why do I need it? The stsTrunkBuilder is a configuration helper utility that creates configuration profiles for running multiple SipToSis and multiple Skype instances. It creates channel startup/shutdown scripts and Asterisk compatible peer information. It creates the chaining configuration of multiple Skype accounts so you can advertise a single Skype Userid and still take multiple incoming Skype calls. stsTrunkBuilder makes multi channel setup and making changes much easier. If you intend to setup only a single SipToSis channel then you do not need it. Get the stsTrunkBuilder for Linux or stsTrunkBuilder for Windows

What SIP phone adapters can I use with this SIP Skype Gateway? There are many SIP phone adapter devices that will work. The Linksys Sipura devices have a flexible dial plan that allows setup without a PBX. I use the Linksys SPA3102 since I also use a PSTN line. The Fritz!box is another powerful device. If you aren't using a SIP provider then just about any SIP device should work. It also works with many softphones such as X-Lite and XPRO. Linphone does not work at this time.

For an ATA to directly connect to SipToSis without a PBX using the same line as a SIP service provider you must meet the following requirements:

• The ATA must not be configured with an outbound proxy. (Some SIP providers require this)
• The ATA must be able to route calls via the dial plan. Linksys and Fritz!box devices can do this easily.

How do I setup a PBX such as Asterisk to do Skype trunking/connect Asterisk to Skype? You can setup a PBX to make multiple active Skype calls by setting up multiple instances of Skype and SipToSis. Each instance/channel will need it's own configuration profile. This is easily done using the stsTrunkBuilder. It can create all the individual SipToSis configurations for you as well as make scripts for starting and stopping the trunk. You can set it up on the same box as your PBX or separate. See Skype Trunk Setup Guide for setup instructions.

What SIP PBX's will it work with? It is known to work with plain Asterisk, AsteriskWin32, PBX-In-A-Flash, Elastix, Axon, trixbox and 3CX. It should work with almost any SIP PBX.

How many concurrent active Skype calls/channels can I make using this SIP Skype gateway with a PBX such as Asterisk?
You are limited by your available resources (memory, bandwidth and cpu). Here's the approximate per channel memory consumption and cpu usage for both supported trunking platforms.
Windows - 56-70mb - Approx 12% cpu with an AMD Athlon 2700
Linux - 52mb+ for SipToSis & Skype and 16mb+ for Xvfb - Approx 5% cpu with an AMD Athlon 2700 using snd-dummy. If you use a different method to create X sessions, your memory usage will be different.
Conservative Ram Calculation: (#channels x 100MB) + OS ram requirements MB + Other APP requirements MB = TotalRam MB
CPU usage above is with standard PCMU codec (30ms packets) and RFC2833 DTMF. You will have to experiment to find the limit for your environment. You can always expand your channel count by adding more boxes. There are no built-in limits. Use of a broadband internet service provider with sufficient bandwidth is required for multi channel operation.

I have run 10 channels myself - I do not have enough bandwith to run more. Some have run 40 channels. If you run 40+ channels I recommend a dedicated 64 bit quad core server with 8gb ram running linux.

If you plan on using Skype's Subscription plans, please respect their Fair use policies. Using multiple subscription plans in a multi channel setup will likely violate their policy. Please contact Skype directly regarding this.

What Skype client versions does this SIP to Skype Gateway work with?
For Sip to Skype on windows, at least 3.x. I recommend 3.6-3.8+ and 4.0.0.215 or higher. Callback and conference were tested with 4.1.0.141 version. 4.1.x version will not leave a voicemail message due to a Skype bug. 4.2.0.141 works with voicemail although Skype may hang or crash if you cancel during the greeting.
For Sip to Skype on Linux, it is known to work with 2.0.0.72 and 2.1.0.81 (2.1.0.47 version has issue with multiple incoming Skype calls - multiple outgoing works fine). Two stage callback with 2.1.0.47 with Skype as originator using Skype outbound does not work. 2.0.0.72 and 2.1.0.81 have not been tested with this feature. Conference calling with 2.1.0.47 has audio problems (at least when using snd-dummy). Again, 2.0.0.72 and 2.1.0.81 have not been tested with this feature. Static versions are recommended to avoid memory usage problems.
For Sip to Skype on MAC OS X - 2.7.0.330.
It may work with other versions as well.

I want to use this with Asterisk/PBX, does it have to be on the same computer? You can put it all on a single box or split components. The only restriction is that Skype and SipToSis have to be together on the same computer. For many channels, I would recommend that Skype/STS be on a separate computer (especially windows) than your Asterisk box, but this is not a requirement.

How is the performance with multiple active calls?
Linux CPU usage under varying conditions.
These shots are in a virtualbox (not an ideal setup) in 32 bit mode using a Vista 64 Host Dual Core AMD Athlon X2 4850e.
Skype sound device setup is a 2 card snd-dummy configuration. SIP codec for all channels is PCMU.
6 calls were from the same Vista host using softphones. The other 4 were made from another PC using softphones.
The Ubuntu distro is a 32 bit 9.0.4 (desktop) with 1.5GB total ram allocation. Approx 800MB was used for the entire setup.
10 Idle channels - There really are 10 they just are not "top" processes.
10 Active channels to echo test - Despite the apparent high load, there was no issue.
6 Active channels to echo test
I will attempt to do similar tests on a dedicated linux box when I have more time.

Windows Vista CPU usage under varying conditions.
These shots are on a Vista 64 Host Dual Core AMD Athlon X2 4850e.
Skype sound device setup using on-board sound device. SIP codec for all channels is PCMU.
6 calls were from the same Vista host using softphones. The other 4 were made from another PC using softphones.
10 Idle channels
10 Active channels to echo test - This system maxed out at 8. It struggled to get 10 going.
6 Active channels to echo test

What should I know about the Callback and Conference features?
There are some compatability issues with certain Skype clients. Windows client 4.1.0.141 works well with both. If making multiple PSTN calls via Skype, you will be charged for each outbound connection according to Skype's billing system. In the case of a Skype subscription plan, each PSTN connection will use up time. Calls to Skype users are always free. SIP side supports a single connection. You can use a SkypeIn/Skype Online Number/SIP DID/Skype User/SIP User to initiate these calls. You can combine a single SIP and multiple Skype targets (Skype or PSTN). In the case of a DID, it must support callerID. Ask your provider directly if your country supports caller ID. Callback is not legal in all countries, check it here.

Can you show a call flow example of how this SIP Skype gateway might work with a PBX.

• SIPCaller --> Asterisk/PBX --> SipToSis --> SkypeSTS --> SkypeUser or SkypeOut to PSTN
• SkypeUser or SkypeIn --> SkypeSTS --> SipToSis --> Asterisk --> SIP destination

Can you show a call flow example of how this Skype to SIP Gateway might work with a SIP device.

• SIPDevice --> SipToSis --> SkypeSTS --> SkypeUser or SkypeOut to PSTN
• SkypeUser or SkypeIn --> SkypeSTS --> SipToSis --> SIP Device or SIP Destination

Can you show a call flow example of how this Sip to Skype Gateway might work with a SIP provider.
SipToSis would have to register with the SIP provider to take incoming SIP calls from provider.

• SIPCaller --> SIPProvider --> SipToSis --> SkypeSTS --> SkypeUser or SkypeOut to PSTN
• SkypeUser or SkypeIn --> SkypeSTS --> SipToSis --> SIP destination

What keeps unauthorized people from using my SIP or Skype accounts. That is totally up to you. If you expose the SIP end to the internet, you can control access by callerid, realm and IP Addresses/blocks combinations. You can also setup PIN authentication/dialing by callerid, realm and IP Addresses/blocks combinations. For incoming Skype calls, you can control access by Skype User or setup PIN authentication/dialing by Skype User.

How do I call a Skype User when I can only dial numbers on my phone? SipToSis has powerful mappings and transforms to help with that. You can also translate numbers to Skype users in your PBX or Skype client if it has speed dial support. See the trouble shooting page for info.

Do I have to run the Skype client to use this Skype SIP Gateway? Yes. This program does not bypass the Skype client. All Skype Gateways operations are performed using the Skype API which means SipToSis must be run on the same computer as Skype.

Where is the source code? For GNU GPL distributions, the source code is included in the distribution download. This way you get the correct source for a particular distribution with no version confusion and satifies the GNU GPL source availability requirement.

All this looks too complicated for me. Isn't there an easier way?
Click Skype Calls Without a Computer to see devices that can use the Skype service without a computer or software.

I have seen this project called Sip-To-Sis, Sip-ToSis,SkypeToSis,Skype-To-Sis,Skype-ToSis,SkypeTo-Sis,SkyToSis,Sky-ToSis and SipTo-Sis. What is the official name?
The official name is SipToSis as shown on this site.