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.

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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.

Have you considered building a downloadable app which would run in VMWare Player?

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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.

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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.

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