Call Forwarding with your PC

[ravenmaddox]

I have this idea. I want to put the modem back in my PC and have it set up to answer the analog line, dial out voip to my cell phone, and connect the call, with as little break or pause as possible. Basically call forwarding but so that no one knows it's call forwarding and so that the option doesn't show up on my phone bill.

I'm thinking that maybe I can setup Asterisk to do this, but I've never used it to control an analog line before and I'm not sure if there is a better option out there for what I am wanting. Any ideas?

[nyphonejacks]

you just need a linksys  spa3102

you connect your analog dial tone to the 'LINE' jack...

when someone calls the POTS line it will ring - then you can set up the SIP side of the 3102 to hotdial whatever number you want when it notices ringing on the POTS line connected to the LINE side of the device

the 3102 looks like a pap2 or 2102, but neither of those will work for what you want to do... you need the 3102

[ravenmaddox]

That is perfect! Thank you. I'm glad I asked... $70 is a lot cheaper than all the time and effort of setting up asterisk again (which I prolly will eventually anyways to save on future spoofcards) I only hope that there is no audible click or delay that would clue the caller into the fact that the number is being routed and redialing.

Was reading thru the config documentation and my eyes are going crossed after hours of reading thru old PLA posts, but this looks perfect. All that remains is implementation.

[nyphonejacks]

there is no indication to the caller, in fact your phones connected to that line will still ring at the same time that the VoIP line is calling out and ringing whatever phone you have it hotdialing to... so you could either answer on your home phone, or the phone that it is hotdialing..

[ravenmaddox]

Awesome... I just put the order thru buy.com

*EDIT* I also have to get my list together of the conference lines and some of the direct dials to "All Circuits are Busy" etc recordings... Hrmmmm...

[nyphonejacks]

i do not use conference lines much.. but other than talkee.com magicjack and nettalk both provide free conference numbers..

as for the recordings, i would check thisisarecording.com

[ravenmaddox]

I have a phone plugged into the 3102 from which I can pick up and dial any number, routing thru my voip.

I can call my analog line and after three rings I get a dial tone which is supposed to connect to the voip and allow outdialing, but it doesn't... I've prolly got a bad dial plan here.

But none of the call forwarding works. I've tried having my cell phone call the landline, the 3102 sees is and rings the phone ext, but it nevers forwards the call out over voip as programmed. I must be missing something and I have looked all over the internet to no avail.

Any other ideas?

[nyphonejacks]

i know that there is a dial plan to make it work the way that you need it to, because i have set up many door phones like that with the 3102, but the guy in the office where i work usually sets up the dial plans for me, i could try to find out for you tomorrow sometime when i am at work

[ravenmaddox]

Cool. I'd definitely appreciate it. I won't be back home until Friday night so no rush till then. I think I found something online about hotdialing and setting up the dial plans but I'm on my phone right now and can't read the PDF. I really appreciate all the help. The 3102 is definitely more easy than asterisk 

Also do you know how to make it set a caller I'd instead of unknown? I'm using link2voip and I know how to set it thru asterisk but this device has me overwhelmed at the moment yet in awe of all the options. Thanks again.

[nyphonejacks]

will send an email to my work email so that i can remember to ask these questions to the guy who deals with provisioning these boxes and will try to get an answer for you by Thursday (the guy works from home on Friday)

hopefully i will have some time in the office before then, since i have a few installs scheduled for this week..

[ravenmaddox]

I got it all figured out, finally!

I am able to set callerID thru Link2Voip by creating an additional SIP username with whatever ID I want and set "LINE 1" to register using those credentials. I can't change it on the fly as I can with Asterisk tho, which means eventually I'll be setting up a dedicated box, but this will suffice for now.

To get the HotDialing set up, I mostly followed the instructions here with the addition of changing some FXO Timer Values

PSTN Answer Delay: 3 <--- gives just enough time for caller ID to pass, setting to zero barely saves 1 sec off total dial time
PSTN Dialing Delay: 0
PSTN Ring Thru Delay: 0 <--- so that my attached phone rings as quickly as possible

I have the analog line forwarded to my mobile phone, and then changed all my mobile phone call forwarding to various numbers that answer with "Due to Network trouble..." etc. from This is a Recording

My next project is either to hack the firmware or find someway to pass a variable to change id on the fly.