We have a really specific problem in the office and I would like some opinions on how to solve it the best using e.g asterix.
The ultimate goal would be to have all incoming calls on our landline forwarded to my workstation (over the LAN) and I would use my headset as a telephone.
I would like your guys take on the plausability of that. Is it even possible ? what are the caveats ?
I guess it would be like skype just using the office number.
I guess The more realistic goal would be to connect the current PSTN-port to asterix, have asterix forward the calls to an IP-phone on the LAN next to my workstation.
I have a via epia M2 1ghz machine to use for it running debian, what other hardware would I need to connect to the PSTN´s rj11 port ?
What are my options ? and will asterix even solve my problem ?
Currently my office and the buildings only PSTN jack is located far from eachother, thus the need for forwarding calls to the network
You can use one of the below ATA’s FXO port to connect to the PTSN and then send the call to a SIP client on the network weather that be a soft phone on you computer using your headset or a IP phone.
External ATA with FXO = Linksys SPA2102
Internal FXO Card For the * Server = Digium TDM401B
[quote=“root52”]That should not be a problem at all.
You can use one of the below ATA’s FXO port to connect to the PTSN and then send the call to a SIP client on the network weather that be a soft phone on you computer using your headset or a IP phone.
External ATA with FXO = Linksys SPA2102
Internal FXO Card For the * Server = Digium TDM401B
There are others out there but that is a start.[/quote]
Thank you for you answer.
Sounds really good, correct me if i´m wrong. -But isn´t the linksys box used to connect my old POTS telephone to the network ?
Currently i´m using an old wireless dect telephone connected to my analog PTSN (which is really a linksys VoIP router P-2602HW-D1A supplied by my ISP)
This is for obvious reasons less than ideal, thats why I would like to convert the analog PTSN jack into something network based.
I take the Internal Digium TDM401B could solve that in symbiosis with asterix ? Although 4 ports seems a bit overkill, are there any single-port alternavtives ?
Without knowning my ISP, is there any way to just forward SIP-based traffic to my LAN instead of effectivly terminating it in the linksys VoIP router P-2602HW-D1A. It´s an ADSL2+ connection.
I have asked my ISP about this earlier and of course the answer was just NO!
The goal still is to have my office number forwarded to a soft-phone SIP-client on my workstation on the LAN. And being able to call-out aswell.