still finding my feet with asterisk at the moment…
setup is voip phones and asterisk machine all on an internal lan.
when one phone calls another and two people converse, does the voice traffic go ‘through’ the asterisk machine or does it just go direct from phone to phone? and can asterisk be tweaked to do either scenario?
Depends. Depends.
If you don’t use any features that require:
- transcoding;
- monitoring/recording of the speech path;
- combining the audio from several sources;
- detection of DTMF digits;
and you have the directmedia parameter set to yes (for a network with no NAT affecting the call);
and the phones accept re-invites to direct media,
the RTP for an established all will go directly between the phones.