So I have a spa2102 (SIP -> 2 FXS ports) that talks to the plain old telephones in my house. Line 2 is set to use a voip provider (callcentric, as recommended by mlug!) so far so good. What I would really like to do is have some sort of routing where a little server does voicemail and forwards it to my email account. I'd also like laptop softphone and home phone both ring when someone phones home, and be able to have glorious voice menus to allow per person extensions and distinctive rings, and ... I dunno what, but it'll be fun.
My router/firewall is debian, and apt-get install asterisk worked, but the configuration looks really complicated. It doesn't seem to have any getting started type documentation, just sort of reference material. Since I have the SPA box, I don't actually need any POTS/PSTN function. without any PSTN to do, do I even need asterisk? Is there pure SIP-based switch that would be easier / faster / simpler ? I have the ekiga softphone on my laptop, and an asterix prompt... I guess the first thing I'd like to do is create an account on asterix for the ekiga phone, and another one for "line2" on the SPA, and phone one from the other... (leaving line1, the real phone, alone for now.) Do I just mess with sip.conf for that? decent "getting started" type guides?
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