I’m getting my first Asterisk server up and running and am trying to understand exactly how sip.conf works.
I have a VoIP handset and a phone number with engin (Australian VoIP provider). I have a few questions:
1.) What does “register” achieve in sip.conf? Does it simply subscribe my Asterisk box to incoming calls from my SIP provider? Is it necessary?
2.) It seems that “best practice” is to set up 2 channels for my VoIP provider (incoming and outgoing, i.e. user and peer) and then 1 channel per handset (i.e. friend). Yet I’ve seen sample configs which only use one SIP channel, type=friend, to deliver the whole lot. Presumably in this latter situation, the handset connects to Asterisk with the same credentials Asterisk uses to place the call with the VoIP provider. Is this the only reason this 1-channel setup works for people? What is the best way to handle this?
3.) Assuming I create separate channels for all of these, am I just creating unnecessary dialplan headaches for myself?
Looking forward to your thoughts!
Thanks, and sorry if this has been addressed already. Google wasn’t being my friend