I’m new here and i’m looking for simple tutorial/config steps for converting a analog phone line to VoIP phone line, so I can use VoIP phone to make calls instead of analog phones.
I currently have a SANGOMA AFT-Remora A200-R with one FXO module. When I can get it working I plan to buy a VoIP phone, for the time being i’m using SIP client on my PC.
Why is this a FreePBX question? A simple configuration would use an FXO gateway box.
Even if you have an Asterisk compatible card, I’d argue that a simple configuration for a single line pass through would be better done with bare Asterisk. If you need detailed instructions to do it with FreePBX, you have negated the possible advantage of using FreePBX, in that it might be slightly easier for an inexperienced user to configure.
I got the card from a sale, I want to use it to convert the phone line to be VoIP because I’m sending ethernet to another location with bridge antennas and I want my phone to be there.
So you are suggesting that it would be better done with bare Asterisk instead of using FreePBX.
If so, could you provide some useful information that I could use to get that setup.
The reason I say bare Asterisk would be easier is that the context for outbound calls is likely to be just (note not tested):
exten => _X.,1,Dial(DAHDI/1/${EXTEN})
and that for incoming calls
exten => s,1,Dial(PJSIP/myphone)
compared with the 100s of lines that FreePBX uses.
The only thing that is likely to cause difficulty on FreePBX is the DAHDI configuration, but that will be a similar issue on bare Asterisk, and the bare configuration is much closer to what people would really want for debugging it.
Getting detailed information for configuring an obsolete card is likely to be difficult, as the vast majority of FreePBX users use SIP. You are more likely to find people with a hobbyist type interest in them on the Asterisk forum, and they will expect bare Asterisk configurations.