Asterisk

Asterisk
Participants	Folkert
Skills
Status	Active
Niche
Purpose
Tool	No
Location	Space
Cost
Tool category

Asterisk

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