39 lines
1.2 KiB
Plaintext
39 lines
1.2 KiB
Plaintext
;
|
|
; /etc/asterisk/pjsip_users.conf
|
|
;
|
|
; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
|
|
;
|
|
[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 = IfYouTolerateThisThenYourChildrenWillBeNext!
|
|
|
|
; 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
|