I HAVE A DREAM Incredible PBX 11-12 and SIAF (Skype integration)?

w1ve

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Thanks to the great work Ward and Team have done, I have a nice Incredible PBX 11-12 running in a DO Droplet.

I'd love to integrate Skype... However, the SIAF instructions are pretty old. Will it work as is?
Is their anything that I need to change?

Thanks!
 

wardmundy

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M$FT killed Skype support on the Linux platform a long time ago. Sorry. Best to buy an Android-based SIP phone (Grandstream) that includes Skype as a separate Android component.
 

sukasem

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New version of Skpe does not work with oss sound anymore. It require sound device and I can't find a way to use dummy sound on linux yet. So, I have to run Windows on the cloud and use virtual sound software (virtual cable) and run FreeSwitch on windows machine and forward call to iPBX server.
 
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Maybe with newer Ubuntu versions we might be able to get it working again. . I always wanted to ness around with it before but never had a chance to test (most I mean free time). So I'd be curious to try and help some if needed or wanted.
 

chris_c_

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M$FT killed Skype support on the Linux platform a long time ago. Sorry. Best to buy an Android-based SIP phone (Grandstream) that includes Skype as a separate Android component.

I think what w1ve meant about skype integration might be similar to what's today called "skype for business" and "microsoft lync" and "cisco unified communications"...

1) when you have bought a SkypeOut subscription (US$2.50/month for 10,000 minutes to call from skype app to landlines and mobiles in the US and Canada = this is using skype's proprietary call termination protocol similar to sip termination), setting skype up on the same machine as the pbx server so that asterisk could bridge its outgoing calls thru to the skype app, for outgoing sip termination! And also, to possibly have the incoming skype calls ring thru to the pbx and be handled by the asterisk incoming call rules...

2) be able to "call out" from asterisk pbx, directly to "skype usernames"... which don't necessarily have a phone number.. simply by bridging over to the skype app running on the same server..

3) dial by name using the skype app's people directory... you can save tons of usernames and numbers in the skype directory inside the skype app..

4) video call bridging... using the popular nortel 1535 color sip video phone, make it so that you can call out of the nortel, thru asterisk pbx, over to the skype app running on the server, out to a skype username, and have the video call just work. This would have to translate the video protocol from sip video call to skype video call format..
 

w1ve

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Sorry.. .been away and wanted to get back to this thread. I was thinking of SIAF -- Freeswitch running mod skypeopen on 127.0.0,1. I thought that skypeopen still works on
Freeswitch, no?
 

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