Hi,
When I use autocreatepeer I get one way audio from pots back to sip. The SIP phone can hear the POTS phone, but not the reverse. Since I am new to this, I am still trying to figure out what I am missing. I am inside on a LAN, no NAT’ing involved. PBX and SIP phone are on same subnet.
Below, is a snippet from my sip.conf. If I use in this manner, I don’t have the 2 way audio problem. If I use autocreate peer, I only get one way audio. What’s missing?
As an aside, I can’t find any docs on autocreatepeer. Are there other features to this option? For example, if I want to create 20 phones, can I use autocreatepeer, but ensure the user configures a username and secret? Thanks!
[82010]
type=friend
context=incoming
;username=chicago
;secret=snom
host=dynamic
dtmfmode=auto
canreinvite=no
nat=no