I’m quite new to Asterisk (been “using” it for a couple days now), and I’m trying to configure a system for my company.
I currently have about 10~ ish VoIP lines (connected to Grandstream routers) and 2 analog lines.
I already made my order for a TDM400P, and my main problem is regarding the VoIP lines.
I don’t know how to connect the VoIP routers to the server, or if I should connect directly RJ45 to a switch togheter with the server.
Currently, this is my main problem.
Thanks in advance.
PS: I’m reading Asterisk TFOT atm, page 79.
The Grandstream “routers” I mentioned before, are HandyTones, not sure of the model as they dont say anything below.
Your grandstream devices are called ATAs, analog telephony adapters. They connect to a SIP account over IP and act as a gateway allowing the FXS port access to the SIP services.
You should connect the ATAs to the server with an ethernet switch, and connect your internet router to that also. (or just plug them all into your existing switch). Now in asterisk’s sip.conf you must define each of the IP lines, as well as define accounts for the handytone’s. Now go to the handytone’s config and tell them to register to *. Lastly, setup the dialplan (extensions.conf) so the calls are managed…
keep reading TFOT
Actually, my idea is to -remove- HT’s directly. By using SoftPhones and letting * to log on my ITSP.
Still, your comment applies.
By the way, does sip.conf users moved to users.conf ?
as i understand it users.conf is a quick way of setting up a sip/iax acct, mailbox, etc. I think you can use either or, just users sets up more stuff at a time