Asterisk: Difference between revisions

Revision as of 11:51, 29 October 2022

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

Erratic 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
3010: Tahtkev (Softphone)
3011: Tahtkev (Deskphone)
4001: bar
4222: rookhok
4333: studio
4455: zaal1
4666: zaal3 (gamershok)
4667: zaal2
9999: party line (conference call)
4200: buZz
1990: boots
4500: zarya
4501: zarya lte cell (add local lte nr behind 4501)
4300: niz

6669: ????
1234: ????
2001: flunk
4999: 'tester'
4700: r3boot (deskphone)
4701: r3boot (softphone)
6002: ????
9100: ????
9200: ????
4421: ????
1515 Melan
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()

Getting Access

It is possible to link your own PBX to the NURDspace PBX. This will make it possible that you can call extensions at NURDspace, and that people at nurdspace can contact you at home. To get this to work, you need to request a vpn tunnel together with a SIP account, and follow the documentation to install and configure your very own PBX.

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
~Various Poly/Polycom VVX series phones

@@ Line 45: / Line 45: @@
 * 1990: boots
 * 4500: zarya
+* 4501: zarya lte cell (add local lte nr behind 4501)
 * 4300: niz