1.9 KiB
Voice over IP
Consists of the following components:
- Asterisk server at asterix.srv.it-syndikat.org
- HT802 analog phone interface connected to Bunkertelefon
- Homeassistant VoIP integration
- SIP trunk to EPVPN
- SIP trunk to PSTN via sipcall
Asterisk
Runs on asterix.srv.it-syndikat.org
.
Central PBX, talks to all other endpoints. Configuration happens mostly in /etc/asterisk/pjsip.conf
and /etc/asterisk/extensions.conf
.
CLI can be accessed using sudo -u asterisk asterisk -r
. Useful commands:
- reloading:
reload
to reload everything,pjsip reload
/dialplan reload
for partial reloads - delete stuck SIP registration:
database show registrar/contact
, then e.g.database deltree registrar/contact 6002;@332c500bfb09158a3a3a9ef53913cd6a
- logging:
pjsip set logger on
to show SIP packets - dialplan help:
core show applications
/core show functions
EPVPN
Asterisk is registered in EPVPN on extension 1754. Outbound calls to EPVPN are possible with prefix 9, inbound calls go to Bunkertelefon.
sipcall
We have a prepaid sipcall phone number, +43 720 519629
. Outbound calls to numbers starting with 0
are routed though here to PSTN. Inbound calls go to Bunkertelefon.
HT802
Analog Telephone Adapter for Bunkertelefon, registered on extension 6001.
Web interface on http://ht802.asozial.it-syndikat.org, credentials in Vaultwarden. Has a machine-friendly-ish SSH interface too.
Config export in voip/ht802/
directory.
Picking up and not dialling for 5 seconds automatically connects to Homeassistant.
Homeassistant
Native VoIP integration, registered on extension 6006.
Call deterrence
Because the Bunkertelefon is quite loud, whenever isitopen is closed, callers will first be greeted by a GLaDOS recording telling them to go away. Pressing 1 will make the phone ring anyway.