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Do you remember my blog post "A SIP address for Skype? Better the other way around!"? This mission has now been accomplished. As of yesterday you can call me on Skype and I will answer this call on my desk phone or cell phone using SIP VoIP telephony. As I always try to achieve, this is a totally free solution.

I have joined Voxeo's developer program for their Evolution application, a visual design tool for interactive voice response (IVR) systems. Part of the deal is that you get a strange phone number with a +990 country code. There is no country associated with this code and Skype users can call these numbers for free. My Skype account is now being forwarded over Voxeo to a SIP address from Gizmo Project which I manage on Voxalot to make use of it's call connection rules and voice mail.

Have a peek on my settings:

A better explanation can be found at the Voxeo support forum. I wonder what VOIPSA's Dan York would say. In January he started a discussion with his blog post "Skype says "No" to VoIP interoperability - *because customers aren't asking for it!* - Well, I am!". He is, by the way, working for Voxeo and this partly solution for his problem comes from his own company. So I guess he was always aware of this trick.

I am happy now that people can call me with Skype and I don't have to keep me computer running or buy a special Skype phone for this purpose. That's the reason why I nearly never used Skype. I don't like applications which keep me tied to my computer in order to receive messages or phone calls, like Skype or the MagicJack normally do. Let's see which other solutions I can develop with Voxeo. Their visual tool makes the design of VoiceXML fairly easy.

Labels: free phone calls, Gizmo Project, Gizmo5, MagicJack, SIP, Skype, VoiceXML, VoIP, Voxalot, Voxeo

                                      

comments:

i am happy you finally figured this out. i have had a similar setup for several years now. has always worked great.

# posted by  Anonymous : Monday, 31 March 2008 08:13:00 o'clock CEST

 

Markus,

i very interesting hack and one that i plan to develop when i have a bit of time would be to write a small bit of callXML code that could be hosted on evolution that automates the entire process of using the 'ether hack' that was recently posted on viney's 'voipguides' blog. this would result in a SIP URI, a skypename and regular landline phone number that rings practically any phone on earth landline or mobile. at no cost to the caller of recipient. this could be very easily accomplished with a small bit of callXML coding. of course you could also use callback solution for outbound calling as well.

# posted by  Anonymous : Monday, 31 March 2008 08:38:00 o'clock CEST

 

will it be possible to use voxalot sip address than gizmo?

# posted by  Anonymous : Monday, 31 March 2008 10:35:00 o'clock CEST

 

Of course it should work with Voxalot too.

Just put "sip:123456!198.65.166.131@sbc-staging-interal" in the phone number field, where "123456" is your Voxalot number and "72.51.47.59" the IP address for voxalot.com. I haven't tried it yet, but this is a solution according to the Voxeo forum.

If it doesn't work, try "64.34.163.35" for us.voxalot.com or "202.60.88.238" for au.voxalot.com or "85.17.19.194" for eu.voxalot.com.

Good luck!

# posted by  Markus Göbel's Tech News Comments : Monday, 31 March 2008 11:01:00 o'clock CEST

 

im back.its working fine now with voxalot. just that i have to edit the 'interal' as i copy/paste, i totally overlooked it.
thanks for this blog..as usual ;)

# posted by  dc : Friday, 4 April 2008 05:17:00 o'clock CEST

 

This post has been removed by the author.

# posted by  vadim : Thursday, 17 April 2008 12:23:00 o'clock CEST

 

I have, sip: 123456!198.65.166.131@sbc-staging-internal at the Forum Voxeo support forum prompted internal to correct interal but is not running what part overlooked. skype put in preadresatsiyu +99000936 99960XXXXX how to fix it?
Regards

# posted by  Anonymous : Thursday, 17 April 2008 12:27:00 o'clock CEST

 

I have, sip: 123456!64.34.163.35@sbc-staging-interal . but is not running what part overlooked. skype put in a redirect on +99000936 99960XXXXX as correct?
Regards

# posted by  vadim : Thursday, 17 April 2008 12:50:00 o'clock CEST

 

After reading your blog, I registered and tried to create application like yours from skype to SIP.In the transfer step, I gave phone number as my gizmo5 SIP account and I chose type as Bridge.

My SIP URI is sip:1747xxxxxx@proxy01.sipphone.com and I entered sip:1747xxxxxxx!198.65.166.131@sbc-staging-internal
(as per forum) Is this correct?

To test this app, When I try to call from 1-800 number with PIN as well as from skype,

I get a message saying that "Unfortunately the system is not currently available for use. Please try your call again at another time". Am I missing anything here ?

Also I read online somewhere that I need to email voxeo support team to issue a “token” that I can use for outbound calling from my app. Do you have to do this?

I sent opened a ticket with voxeo support, they came back with this message:
"
Due to the way Skype passes caller ID, performing a transfer to a SIP destination will fail. Engineering is currently working on a workaround to correct the issue,
but at the time I don't have an ETA on this. You can, however, perform a transfer to a SIP destination if you call in via SIP or a standard phone.
"
Any ideas? Can you please explain the steps you followed in detail?

# posted by  SaRa : Friday, 25 April 2008 04:22:00 o'clock CEST

 

I forward my Skype calls over Voxeo with "sip:1747XXXXXXX!198.65.166.131@sbc-staging-internal" and it works great. Just a minute ago I checked it with a call to my Skype address which got forwarded to my cell phone.

The "token" for outbound calls is not necessary for that since calls from Skype to your +990 number are free.

I remember also getting the message "Unfortunately the system is not currently available for use. Please try your call again at another time" when I tinkered with Voxeo. I had to always hear it when the call forward was misconfigurated.

Here I copy the most important details for you:

Transfer to: Phone Number
Phone number: sip:1747XXXXXXX!198.65.166.131@sbc-staging-internal
Type: Bridge
Where would you like to send your callers in the following instance(s)?: Hangup

# posted by  Markus Göbel's Tech News Comments : Friday, 25 April 2008 12:09:00 o'clock CEST

 

After following the above steps(support team has enabled SIP outbound dialing on my account)it seems to work sometimes and sometimes its not. Now it does not work at all. skype call just hangs up. nothing happens. Did you experience this problem? What needs to be done to fix this?

Also when I signin to my voxeo evolution applocation on their website, I see that the Deployment Type is "Staging". what exactly is that? Do I need to deploy my app?
How long my application will be in voxeo evolution for free? Do we need to pay or anything?

How did you setup gizmo5 to call your phone? Are you using landline or the mobile? I am in US. I'd appreciate if you give some details.

# posted by  SaRa : Wednesday, 14 May 2008 05:38:00 o'clock CEST

 

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