pjsip.conf

[global]
type = global
endpoint_identifier_order = username,ip,anonymous

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:16555
tos = cs3

[transport-tcp]
type = transport
protocol = tcp
bind = 0.0.0.0:16555
tos = cs3

; If you don't have TLS certificates and aren't using TLS, don't uncomment these. PJSIP will error out and not load.
;[transport-tls]
;type = transport
;protocol = tls
;bind = 0.0.0.0:16556
;tos = cs3
;cert_file = /etc/letsencrypt/live/example.com/fullchain.pem
;priv_key_file = /etc/letsencrypt/live/example.com/privkey.pem
;verify_server = no
;method = tlsv1

; PJSIP lets you set up multiple transports, so you can run different versions of TLS on different ports so supporting equipment can run TLS 1.2+ and you can still have TLS 1.0 on a different port for backwards-compatability. e.g. GS HT802 can do TLS 1.2 but HT704 can only do up to TLS 1.0.
;[transport-tls12] ; some older ATAs require TLS 1.0, so also configure a transport with a newer TLS version for those that can support it.
;type = transport
;protocol = tls
;bind = 0.0.0.0:16556
;tos = cs3
;cert_file = /etc/letsencrypt/live/example.com/fullchain.pem
;priv_key_file = /etc/letsencrypt/live/example.com/privkey.pem
;verify_server = no
;method = tlsv1_2

; Sample CallCentric registration for PSTN:
;--
[reg_callcentric]
type=registration
transport=transport-udp
outbound_auth=callcentric_auth
retry_interval=60
expiration=3600
auth_rejection_permanent=yes
contact_user=17775551212
server_uri=sip:callcentric.com
client_uri=sip:17775551212@callcentric.com

[callcentric_auth]
type=auth
auth_type=userpass
password=samplepassword
username=17775551212

[callcentric]
type=aor
contact=sip:17775551212@callcentric.com

[callcentric]
type=identify
endpoint=callcentric
match=callcentric.com

[callcentric]
type = endpoint
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
tos_audio = ef
disallow = all
allow = ulaw
language = en
transport=transport-udp
context=from-callcentric
outbound_auth=callcentric_auth
aors=callcentric
from_domain=callcentric.com
from_user=17775551212
sdp_owner=17775551212
ice_support=no
send_rpid=yes
rtp_symmetric=yes
force_rport=yes
timers=no
--;

[lines-endpoint](!)
type = endpoint
disallow = all
allow = ulaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
inband_progress = yes
tos_audio = ef
device_state_busy_at = 1
trust_id_outbound = no
trust_id_inbound = no
notify_early_inuse_ringing = yes
context = from-internal ; context in the dialplan in which this user originates a call
allow_subscribe = yes
subscribe_context = phreaknet-hints

[lines-aor](!)
type = aor
max_contacts = 1
qualify_frequency = 30

[lines-auth](!)
type = auth

; every user needs to have an AOR (address of record) section, auth section, and endpoint section. To minimize clutter, we use templates for each.
[DeskPhone1](lines-aor)

[DeskPhone1](lines-auth)
username = DeskPhone1
password = samplepasswordhere

[DeskPhone1](lines-endpoint)
callerid = "John Smith" <5552368> ; change the CNAM and caller ID of your line here
;media_encryption = sdes ; for SRTP (voice path), if your endpoint supports it. Make sure to connect to the TLS port so the signaling is encrypted, too.
auth = DeskPhone1
aors = DeskPhone1
;mailboxes = 2368@vmcontext ; for voicemail MWI