Difference between revisions of "Asterisk"

From NURDspace

Jump to: navigation, search
(allowing people from internet to connect to this asterisk instance)
Line 106: Line 106:
 
OR
 
OR
 
* let them connect via [[Wireguard|vpn]]
 
* let them connect via [[Wireguard|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 [[Asterisk/MemberPBX|documentation]] about running an asterisk instance at home.
 
There is also [[Asterisk/MemberPBX|documentation]] about running an asterisk instance at home.

Revision as of 22:28, 16 August 2022

Asterisk
Foon rookhok.jpg
Participants Flok
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"
}
  • 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
 
Our site is hosted by Site4U
Site4U
Our connectivity is made available by BIT
To BIT's website