Just for holiday fun I got my macmini a new incoming Skype line that can be picked up by any extension, transferred to any extension in our Asterisk PBX. The macmini hosts the Skype Client and Asterisk PBX. You could do the same with better results on a Linux server.

Here's the basic recipe:

1)FXO Device
2)USB-Skype RJ11 Adaptor with stable drivers for my OS
3)Recent version of Asterisk on a stable machine
4)Text editor

In particular, I used the following brands/versions:

1)Grandstream Handytone-488 1.0.2.16 for its FXO port
2)CuPhone PPG with TjInit 1.11
3)Asterisk 1.2.1 on MacMini with OS X 10.3.9
4)vi

Because this was my first experience with the HandyTone, I spent a great deal of time banging my head on my desk trying to get the logic (or apparent lack of) of this crucial piece of the puzzle.

The CuPhone USB adaptor just works as advertised right out of the box. I played with it in a number of configurations and if you don't need a pbx, it simply makes your standard favorite analog telephone into a Skype phone. Incoming calls to my extension ring just as usual, as do direct incoming skype calls, ie. the phone rings and you pick it up and start talking. When dialing out, you lift the receiver and your skype contact list pops on the desktop, you highlight the contact you like and press '##' to initiate a call. There is a serious issue with DTMF on outgoing calls, they only work occasionally. But my main focus was accepting direct incoming skype calls. The other irritating thing on outgoing calls is the need to hit '*#*' to bypass skype inorder to make normal calls to other extensions. That's where asterisk comes in handy, let asterisk do the hard stuff...

With all the pieces put together, you could put the macmini in a closet with the screen off. But since it's a macmini, I won't, because it can handle the phones and a whole lot more.I already have a working skype account and working asterisk pbx. Here is the basic setup I have added to receive skype calls to all desks:

* Added the CuPhone's TjInit software and start it just before I startup Skype

* CuPhone is connected to a full power USB port

* Grandstream HT-488 is Powered up

* A phone cord connects from the 'Phone' jack on Cuphone to the 'Line' jack on the HT-488 (FXO port)

* My good ole two line AT&T 732 analog phone is in the HT-488 'phone' jack (FXS port)

* A Cat5 cable connects from the HT-488's 'WAN' port to my network switch

The HT-488 treat's the incoming skype calls as a standard PSTN and transfers (after the minimum 1 ring) calls to my Asterisks 'inbound' context and I have it blast all extensions in the office, to be picked up by the first person available, or the call is sent to the office default voicemail box. But obviously you could do anything you can do with a normal PSTN call.

If anyone would like examples from my working 'sip.conf', 'extensions.conf' or settings for the HT-488, let me know.

If I were doing this again, I'd save $10.00 on the CuPhone by getting the single CuPhone RJ11 Usb Device, and I'd probably spend an extra $20.00 and use the Sipura 3000 instead of the Grandstream HT-488 (seems like there's more support) but I'm set for now. (out of playtime and playmoney)
[/b]

[quote=runningwater13]Just for holiday fun I got my macmini a new incoming Skype line that can be picked up by any extension, transferred to any extension in our Asterisk PBX. The macmini hosts the Skype Client and Asterisk PBX. You could do the same with better results on a Linux server.

Here's the basic recipe:

1)FXO Device
2)USB-Skype RJ11 Adaptor with stable drivers for my OS
3)Recent version of Asterisk on a stable machine
4)Text editor

In particular, I used the following brands/versions:

1)Grandstream Handytone-488 1.0.2.16 for its FXO port
2)CuPhone PPG with TjInit 1.11
3)Asterisk 1.2.1 on MacMini with OS X 10.3.9
4)vi

Because this was my first experience with the HandyTone, I spent a great deal of time banging my head on my desk trying to get the logic (or apparent lack of) of this crucial piece of the puzzle.

The CuPhone USB adaptor just works as advertised right out of the box. I played with it in a number of configurations and if you don't need a pbx, it simply makes your standard favorite analog telephone into a Skype phone. Incoming calls to my extension ring just as usual, as do direct incoming skype calls, ie. the phone rings and you pick it up and start talking. When dialing out, you lift the receiver and your skype contact list pops on the desktop, you highlight the contact you like and press '##' to initiate a call. There is a serious issue with DTMF on outgoing calls, they only work occasionally. But my main focus was accepting direct incoming skype calls. The other irritating thing on outgoing calls is the need to hit '*#*' to bypass skype inorder to make normal calls to other extensions. That's where asterisk comes in handy, let asterisk do the hard stuff...

With all the pieces put together, you could put the macmini in a closet with the screen off. But since it's a macmini, I won't, because it can handle the phones and a whole lot more.I already have a working skype account and working asterisk pbx. Here is the basic setup I have added to receive skype calls to all desks:

* Added the CuPhone's TjInit software and start it just before I startup Skype

* CuPhone is connected to a full power USB port

* Grandstream HT-488 is Powered up

* A phone cord connects from the 'Phone' jack on Cuphone to the 'Line' jack on the HT-488 (FXS port)

* My good ole two line AT&T 732 analog phone is in the HT-488 'phone' jack (FXS port)

* A Cat5 cable connects from the HT-488's 'WAN' port to my network switch

The HT-488 treat's the incoming skype calls as a standard PSTN and transfers (after the minimum 1 ring) calls to my Asterisks 'inbound' context and I have it blast all extensions in the office, to be picked up by the first person available, or the call is sent to the office default voicemail box. But obviously you could do anything you can do with a normal PSTN call.

If anyone would like examples from my working 'sip.conf', 'extensions.conf' or settings for the HT-488, let me know.

If I were doing this again, I'd save $10.00 on the CuPhone by getting the single CuPhone RJ11 Usb Device, and I'd probably spend an extra $20.00 and use the Sipura 3000 instead of the Grandstream HT-488 (seems like there's more support) but I'm set for now. (out of playtime and playmoney)
[/b][/quote]

Sounds great? But a mass market simple to use plug and play solution please for Mr/Mrs Average Joe....

Well done. Definately post all of the details you can.

well, any news on this?

Well, it's actually pretty simple, (after you've pulled your hair out trying things that don't work).

1) Setup the CuPhone with the drivers from their website for MacintoshOSX
2) Set Skype to use the CuPhone USB audio device for input and output.

Now your analog phone will ring when a skype call comes in.

Next step, is the grab that ring before it goes to an analog phone by plugging in an FXO device in place of the phone. I plugged the 'Phone' jack from the CuPhone into the 'Line' jack on the Grandstream HT-488 and put my analog phone into the 'Phone' jack of the grandstream. Without powering up the Grandstream calls to my skype account still ring on the analog phone. Make sure that's true before proceding.

Next step is to make the FXO and FXS ports register to the Asterisk server and direct incoming calls on the FXO (Line) port to the 'inbound' context of the asterisk server, so I can send it to which ever extensions I want to ring. I chose to create extension 2100 for the FXO (line) and extension 2000 for the FXS (phone). On the Grandstream this is done via a web configuration menu. Key here is to match the basic setting 'Forward to VoIP' to the inbound extension name, in my case 'SkypePSTN' (no @blah.blah). I also set the 'basic setting' called 'Number of Rings' to 1 (number of rings before a PSTN incoming call is forwarded to VoIP).

Here are my snips from '/etc/asterisk/sip.conf' on my MacMini.
====== Cut from sip.conf ================
[2100]
type=user ; I set this to user because I'm just doing inbound
user=phone ; not sure if this is correct but it works
username=2100
fromuser=2100
secret=totally
host=dynamic
port=5062
nat=yes
caninvite=yes ; not sure if this is correct but it works
canreinvite=yes ; again not sure, but it works
disallow=all
allow=ulaw
context=inbound

====== cut from extensions.conf ===============
[globals]
DIALIN1=SIP/2222&SIP/2219&SIP/2000 ; extensions that ring on inbound

[inbound]
exten => 3605551212,1,Macro(inbound) ; teliax.com
exten => 7785551212,1,Macro(inbound) ; sixtel.net
exten => 2535551212,1,Macro(inbound) ; sixtel.net
exten => 3605551234,1,Macro(inbound) ; sixtel.net
exten => 8885551212,1,Macro(inbound) ; sixtel.net
exten => 12065551212,1,Macro(inbound) ; sipphone.com virtualnumber
exten => SipPhoneLine1,1,Macro(inbound) ; sipphone
exten => SkypePSTN,1,Macro(inbound) ;skype/ht488fxo

[macro-inbound]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,Playback(Cepstral-Lawrence)
exten => s,4,Dial(${DIALIN1},10,r)
exten => s,5,Voicemail,u${VMBOX}
exten => s,6,Hangup
exten => s,105,Voicemail,b${VMBOX}
exten => s,106,Hangup

====After-Thoughts==========
On linux this might work better using the pci FXO cards from digium, but that isn't an option on a macmini. DTMF doesn't work very well, so this is really an incoming call hack. Outbound works find if you're sitting at the mac mini and selecting your contact with a mouse...

If screen shots from the setup of the grandstream ht-488 would be helpful I can post them as well.