38 lines
1.2 KiB
Plaintext
38 lines
1.2 KiB
Plaintext
; /etc/asterisk/pjsip_wizard_users.conf
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;
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; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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;
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[1000](extension-defaults)
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; Dialplan context name for calls originating from this account.
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endpoint/context = from-ext
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; Voicemail address.
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endpoint/mailboxes = 1000@default
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; Internal Caller ID string for this device.
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endpoint/callerid = 1000 <1000>
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; Username for SIP account. By convention, this should be the extension number.
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inbound_auth/username = 1000
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; Password for SIP account (you can choose whatever password you like).
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inbound_auth/password = IfYouTolerateThisThenYourChildrenWillBeNext!
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; Maximum number of simultaneous logins for this account.
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aor/max_contacts = 1
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; Check connectivity every 30 seconds.
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aor/qualify_frequency = 30
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; Set connectivity check timeout to 3 seconds.
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aor/qualify_timeout = 3.0
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; IMPORTANT! This setting determines whether the audio stream will be proxied
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; through the Asterisk server.
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;
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; If this device is directly reachable by the internet (either by a publicly
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; routable IP, or static port mappings on your router), choose YES.
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;
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; Otherwise, if this device is hidden behind NAT, choose NO.
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endpoint/direct_media = no
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