Asterisk: Difference between revisions

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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 [[Asterisk/MemberPBX|your very own home PBX]].
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 [[Asterisk/MemberPBX|your very own home PBX]].


For space and personal use, we recommend:
* ~Various [[Asterisk/Polycom_VVX_Series|Poly/Polycom VVX series phones]]
[[Category:VoIP]]
We also have a huge stack of phones that need to be put on marktplaats
* ~30 Aastra/[[Asterisk/mitel_6865i|Mitel 6865i]] gbit (~dozen complete kit inc psu)
* ~30 Aastra/[[Asterisk/mitel_6865i|Mitel 6865i]] gbit (~dozen complete kit inc psu)
* ~100 Aastra/[[Asterisk/mitel_6735i|Mitel 6735i]] 100mbit
* ~100 Aastra/[[Asterisk/mitel_6735i|Mitel 6735i]] 100mbit
* ~big box with handsets, cords and feet
* ~big box with handsets, cords and feet
* ~dozen selection of various voip devices
* ~dozen selection of various voip devices
* ~Various [[Asterisk/Polycom_VVX_Series|Poly/Polycom VVX series phones]]
[[Category:VoIP]]

Revision as of 13:37, 30 October 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
  • 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)
    • 1210
    • 1211
    • 1212
    • 1216
  • 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 (please keep ordering of extentions)

telefoons

MAC Address Extension Name
00:04:f2:6f:b1:fd 4001 Bar
00:04:f2:6f:be:7d 4666 Zaal3
00:04:f2:6f:c2:18 4222 Rookhok
00:04:f2:6f:c8:d5 4667 Zaal2
00:04:f2:81;9d:df 4455 Zaal1


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.

provisioning

All polycom phones are managed via DHCP and a http-based configuration provider. See the provisioning page for details.

Filename Description
<macaddress>.cfg Global configuration of phone (needed to let provisioning work)
<macaddress>-web.cfg All specific configuration wrt SIP and customization is set in this file
<macaddress>-phone.cfg All device related settings are set in this file.

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()

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.

For space and personal use, we recommend:

We also have a huge stack of phones that need to be put on marktplaats

  • ~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