Hi All,
We recently converted a server from the chan_sip driver to only pjsip.
Everything is working, but we’re having one way audio issues with an FXO, an Obi 110.
We have this device setup via a Trunk entry. Here’s what we had on chan_sip:
username=8880
type=friend
transport=tcp
secret=somepassword
qualifyfreq=25
qualify=yes
nat=yes
host=dynamic
dtmfmode=auto
context=from-pstn
canreinvite=no
Note the nat and dynamic parts.
I cannot figure out how to replicate this on the pjsip trunk, and therefore we’re having one way audio issues.
I read on this wiki that I need to set the following settings to replicate chan_sip’s “nat=yes”, but I can’t find them in the FreePBX UI under Trunks, only under Extensions, and this device needs to be a Trunk.
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
Anyone know how I can have a NAT’d dynamic host on pjsip?