Hi first post, so hope I'm not violating protocol.
Been using Asterisk as home phone and hobby use for nearly 10 years. I
love this project.
Anyway, would someone mind verifying my pjsip.conf ? This seems to work
well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade
to 14.4.1. Other than that the phone registers properly on 14.4.1.
I can provide a pjsip log as well, but for now I'll start with this.
Asterisk is behind a shorewall firewall on a private natted network. It
has a single interface eth0.
Relevant pjsip.conf:
[transport-tls-nat]
type=transport
protocol=tls
method=sslv23 ;sslv23 enables tls1.2 because reasons
cert_file=XXX ;removed
priv_key_file=XXX ;removed
bind=0.0.0.0:5061
external_media_address=x.x.x.x ;public ip
external_signaling_address=x.x.x.x ;public ip
local_net=192.168.0.0/16
[endpoint-common](!)
type=endpoint
context=users
disallow=all
allow=g722,ulaw,h264
dtmf_mode=info
[endpoint-sdes](!)
media_encryption=sdes
[aor-common](!)
type=aor
remove_existing=yes
max_contacts=1
maximum_expiration=160
qualify_frequency=60
[207](endpoint-common,endpoint-sdes)
;Linphone
callerid=Chris <PSTN number>
auth=207
aors=207
mailboxes=201@default
use_avpf=yes
rtp_symmetric=yes
media_use_received_transport=yes
force_rport=yes
[207]
type=auth
auth_type=userpass
password=supersecretpassword
username=207
[207](aor-common)
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