;
; /etc/asterisk/pjsip_users.conf
;
[1000](extension-defaults)
; Dialplan context name for calls originating from this account.
endpoint/context = from-ext

; Voicemail address.
endpoint/mailboxes = 1000@default

; Internal Caller ID string for this device.
endpoint/callerid = 1000 <1000>

; Username for SIP account. By convention, this should be the extension number.
inbound_auth/username = 1000

; Password for SIP account (you can choose whatever password you like).
inbound_auth/password = MapleCarrotBlueGrainFishSoapSoup

; Maximum number of simultaneous logins for this account.
aor/max_contacts = 1

; Check connectivity every 30 seconds.
aor/qualify_frequency = 30

; Set connectivity check timeout to 3 seconds.
aor/qualify_timeout = 3.0

; IMPORTANT! This setting determines whether the audio stream will be proxied
; through the Asterisk server.
;
; If this device is directly reachable by the internet (either by a publicly
; routable IP, or static port mappings on your router), choose YES.
;
; Otherwise, if this device is hidden behind NAT, choose NO.
endpoint/direct_media = no