| Asterisk | |
|---|---|
 | |
| Participants | Folkert |
| Skills | |
| Status | Active |
| Niche | |
| Purpose | |
| Tool | No |
| Location | Space |
| Cost | |
| Tool category | |
Foon_rookhok.jpg {{#if:No | [[Tool Owner::{{{ProjectParticipants}}} | }} {{#if:No | [[Tool Cost::{{{Cost}}} | }}
general
- ip address: 10.208.11.13
- coherence container: 109
Voip subscription
We have a VoIP subscription from budgetphone.nl. The number is: 0317-794512.
The account from budgetphone.nl can be accessed with username 'penningmeester@nurdspace.nl' and the usual password.
extensions
- 9: main menu
- 1 bar
- 2 buzz
- 3 rookhok
- 4 zaal1
- 5 studio
- 6 gamers
- 8 flok mobile
- not making a choice will ring -all- extensions
- 2222: Music
- 2223: Music skip
- 4001: bar
- 4222: rookhok
- 4333: studio
- 4455: zaal1
- 4666: zaal3 (gamershok)
- 4667: zaal2
- 9999: party line (conference call)
- 4200: buZz
- 1990: boots
- 4500: zarya
- 4300: niz
- 6669: ????
- 1234: ????
- 2001: flunk
- 4999: 'tester'
- 6002: ????
- 9100: ????
- 9200: ????
- 4421: ????
- 1111: flok
- 2992: flok
- 666 (and 6001?): flok mobile
configuring
- /etc/asterisk
- sip.conf
- iax.conf
- users.conf
- extensions.conf
telefoons
- 00:04:f2:ae:68:04 10.208.43.2 Polycom_0004f2 2019-03-15 22:32:16 -NA-
- 00:15:65:86:f4:88 10.208.43.55 SIP-T23G 2019-03-15 22:52:43 -NA-
- 00:15:65:9c:9f:cb 10.208.43.46 SIP-T23G 2019-03-15 23:02:22 -NA-
- 00:15:65:9c:9f:df 10.208.43.49 SIP-T23G 2019-03-15 22:28:29 -NA-
- 00:15:65:a2:2f:c5 10.208.43.32 SIP-T23G 2019-03-15 23:28:31 -NA-
In extensions.conf you configure what happens when someone calls in or out. This is also the place where the /usr/local/bin/push-it.sh script is invoked when someone dials-in.
MPD
audio_output {
type "httpd"
name "pcm wav Stream"
encoder "wave"
port "6666"
format "8000:16:1"
}
- /usr/local/bin/stream.sh
- /usr/bin/curl -s http://10.208.43.94:6666/ --output -
- musiconhold.conf
[default] mode=custom application=/usr/local/bin/stream.sh format=pcm
allowing people from internet to connect to this asterisk instance
- adjust sip.conf and extensions.conf
- add their static ip-address to /etc/iptables/rules.v4
OR
- let them connect via vpn
Then, create a sip account which can be used as a trunk. Add a stanza like below to sip.conf
[member_username] type=peer username=member_username secret=PASSWORD_HERE host=dynamic canreinvite=no insecure=invite,port qualify=yes context=users trunk=yes directmedia=no
Next, create the extension(s) you want in extensions.conf:
; When 4700 is dialed, forward the call to extension 100 running on the PBX of member_username exten => 4700,1,Dial(SIP/member_username/100,90,rum) exten => 4700,2,VoiceMail(member_username@default,u) exten => 4700,3,Hangup()
There is also documentation about running an asterisk instance at home.
member phones
We have a large stock of phones for people to use for voip purposes. You can pick one or more of these to hookup to your very own home PBX.
- ~30 Aastra/Mitel 6865i gbit (~dozen complete kit inc psu)
- ~100 Aastra/Mitel 6735i 100mbit
- ~big box with handsets, cords and feet
- ~dozen selection of various voip devices