Skype To SIP Gateway - Personal Edition
“After trying other Skype-SIP gateways for Linux, I found this one that actually works, so I think I’ll stick with it. Furthermore, I think it is especially clever as it does not try and provide everything but it leverages on powerful, readily available FOSS components, such as jackd, alsa and opal.”
–Carlo Strozzi, CEO of ScriptaWorks Ltd
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This product allows 2 callers, one on Skype and the other with a SIP address, to communicate with each other. By configuring this product and the SIP server (example Asterisk) correctly, you can come up with many interesting working scenarios. |
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The blue and red lines show how calls are routed via the gateway. |
It is possible to do the following:
- Forward Skype callers who are your friends to your mobile number, PSTN number or SIP number.
- Reject all Skype callers whom you do not know.
- Allow yourself and family members with own Skype accounts to call into interactive voice response system provided by SIP server. With this, you can do things like access your voice mails or dial out to PSTN lines.
- Using multiple Skype To SIP Gateways, you can now allow SIP users on different private SIP servers to communicate, using the gateways to provide internet connection via Skype network. This is much like local PBXs of branches of a company connected via internet. Note: This will require multiple PCs, each hosting one Personal Edition of this gateway.
- Allow SIP users on softphones like Ekiga, X-Lite and many others to contact Skype callers directly. The SIP users need not have Skype accounts.
- And many others …
Those who want to try out this product can do the following two things:
- Download and install a copy of this gateway. It offers you 1 minute talk time to test the quality of connection.
- Try an echo test to my gateway. My gateway is connected to an Asterisk server which provides the echo test. Speak when prompted for about 20 secs. The spoken words are then echoed back.
Echo Test
To buy, please visit store.
Here are the screen shots of the gateway:
Alex responds:
Posted: December 2nd, 2007 at 3:37 am →
Hey, that’s probably even greater thing I ever imagined.
Just think about it - you could pump this in your business PBX (say Asterisk) and call skype directly. Ain’t that neat or what??
[reply this comment]
Greg responds:
Posted: February 20th, 2008 at 9:35 am →
Actually it didn’t work for me. It locks up the computer. Try SippySkype at http://www.mhspot.com - it’s free and open source.
[reply this comment]
Lor reply on February 20, 2008:
I am sorry to hear the program locks up the computer. Without specifics from you, it doesn’t help me and others. Your post does help others who may be interested in other offerings, hence I am approving the post.
With a quick browse of your offerings, these are my comments:
- I believe your product is tested for Windows and may not work for Linux. You should consider testing your product under Linux.
- My product is ONLY targeted at the Linux space and is actually written for those who like to have the SIP server and my product running in ONE box. This is savings considering that Windows OS is not cheap.
- Linux has two different sound systems, OSS and Alsa. Skype has made the decision to have all future Skype clients supporting only Alsa. As I cannot find a Skype Sip gateway product built for Alsa, I decided to build this one.
- My product is not opensource because it is difficult to have people donating to support my efforts. It is priced inexpensively to cover overheads to support my efforts.
Without knowing the problem you faced, these are steps that can probably help you:
- Turn “ACPI” off. Some PCs have problems when ACPI is turned on. The XWindows just freezes up.
- if you are not running “low latency” kernel, then for the jackd connection, do not use the “-R” option.
Lastly, Skype client on Linux have problems with Alsa-Jack plugin. I hope they can rectify this soon. The details are in my release notes for my latest version.
Brian responds:
Posted: July 2nd, 2008 at 5:08 pm →
Will you software work:-
1. Is it possible for someone to call a SIP number and then that call to automatically go out on Skype to a specific Skype user?
i.e. Call SIP number, SIP call received but passed directly to the Skype network to be received by a designated Skype user.
2. It is possible for that same Skype user to call out on Skype and access the SIP network as if calling out from a SIP phone?
i.e. A specific Skype user calls out and wants to be able to call anyone on the SIP network.
If you post an answer here then it would be best as my email account isn’t so reliable.
[reply this comment]
Lor reply on July 2, 2008:
For both your queries, they can easily be done when using my gateway together with Asterisk (SIP server). I have Asterisk to handle all the mapping between Skype IDs and SIP IDs. The gateway just do the routing. I believe other SIP servers can be used too, though I have not tried them.
Royce responds:
Posted: September 22nd, 2008 at 2:45 am →
I would like to try your software before buying it. I have tried 2 similar softwares today and both were unsuitable. Luckily they had demo versions.
[reply this comment]
Lor reply on September 24, 2008:
Mine has demo too and you can have talk time of 1 minute without registration. You can also call zhink.com and do an echo test.
farrukh responds:
Posted: September 24th, 2008 at 9:24 am →
Hi
I been working on Asterisk for a while. Do i need another PC to run your gateway or will it run on my asterisk box.
thanks
[reply this comment]
Lor reply on September 24, 2008:
No you don’t. One box for box Asterisk and the gateway. I understand it is a little difficult for some folks to integrate my software given the OS and kernel constraints. So, in the future, I may release a fully integrated stack that includes Asterisk. I have a new release, but is still pondering how best I should release this.
Jb responds:
Posted: September 29th, 2008 at 10:05 am →
I noticed your application seems to have a GUI. Is there any CLI for it? And would it run without the GUI? I want to incorporate it in the same box as a Trixbox, in which it will need to be CLI based since there is no GUI server side.
[reply this comment]
Lor reply on September 29, 2008:
This application is dependent on Skype which uses a GUI. Skype does not offer one without GUI. Should there be a non-GUI Skype in the future, then it is possible to rewrite this gateway as CLI.
Chris responds:
Posted: November 20th, 2008 at 8:04 am →
This will need an edit before it’s posted.
I think, like most people, my requirements are simple. They want to allow people on the dark side to contact them. I have tried other gateways or forwarders and it seems that receiving a call is the easiest thing to achieve but I find the usual implementation is not how I would do it. Take, for example, a Skype user who wants a cheap (free) Skype-in number. He doesn’t want to abandon his Skype chums or pay Mr Skype his outrageous charge for something sip users get for free. So he gets a free sip account which includes a free dial in number (there are many for the UK a few in the US), bogus Skype account to “dial out” to his pukka Skype account and sets up a nice bit of forwarding software to run on a pc in the cupboard under the stairs. The final solution looks like this.
Real phones sip world Gateway Skype world
Joe Blogs –> sip service provider –> sip and bogus Skype -> normal Skype
Joe Blogs picks up the phone and dials your number.
Your sip provider routs the call to your sip device (the gateway)
The Gateway answers the call, opens a Skype connection to your normal Skype client and connects the audio.
You or Joe Blogs hang up and the gateway closes all connections.
Nothing new, but here’s my problem.
Joe Blogs is paying for the call immediately because the gateway answers the call straight away.
Many sip service providers support forking. Many sip devices registered at once. The first device to answer gets the call. That doesn’t give me much of a chance against a PC whose only job is to sit there, digital fingers poised over the receiver, itching to connect my call.
So here’s the the question.
Does your software (or will the next version)?
Detect an incoming call (from Joe Blogs)
Attempt to connect the far side (to your normal Skype)
Wait patiently for you to accept/reject the call.
If you accept, answer the call from Joe and connect the audio.
If you reject, are busy or not online then send a busy signal (or do nothing) someone else will collect the call or call routing may send it somewhere else.
NOT just answer the call before it knows if it can do anything with it.
[reply this comment]
Lor reply on November 20, 2008:
I am not editing your post because I think I can understand what you wrote. Using your example,
Joe Blogs –> sip service provider –> sip and bogus Skype -> normal Skype, let me explain.
Using Vasuntu, see http://zhink.com/site/main/index.php/20081101vasuntu/, which already has the Skype Sip Gateway (SSGWPE) integrated, you can do what you want.
You first has to create a “trunk” under Asterisk to link to your sip service provider. Once that is working, when Joe Blogs dial your sip number, this can be answered by Asterisk which in turn pass the call to SSGWPE which then connects to the final Skype destination. The reverse is also possible.
You should be able to identify who Joe Blogs is by the caller ID. The best way for you is to test what I just said by downloading a CD of Vasuntu and try it. The instructions are in the CD.
Chris responds:
Posted: November 20th, 2008 at 8:16 am →
Have you considered building a downloadable app which would run in VMWare Player?
[reply this comment]
Lor reply on November 20, 2008:
No, because I think if Vasuntu is supposed to be a “server”, there is no reason to run it under VMWare.
Saleem responds:
Posted: June 29th, 2009 at 7:46 am →
Hi,
I like to use your product if your product supports the following thing. Because everybody is talking about calling the skype user from SIP. I want the reverse one.
I like to call a SIP user (my friend, who doesn’t have all these asterisk kind of softwares) from my skype software. Instead of using a softphone like x-lite to call my friend, I like to use the skype network which gives better clarity of calls. After installing your software, how can I dial the SIP no from my skype’s dialpad?
Thanks.
[reply this comment]
Lor reply on June 29, 2009:
If you want to use the Skype network for your communication, then you have to install the gateway over at your friend’s place. When you call your friend, it goes from Skype to Skype (internet) and then forwarded to Sip (local/internet).