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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. | 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. In order to do this, edit the following files: | |||
{| | |||
!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 === | === MPD === |
Revision as of 12:27, 30 October 2022
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
- 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. In order to do this, edit the following files:
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.
MPDaudio_output { type "httpd" name "pcm wav Stream" encoder "wave" port "6666" format "8000:16:1" }
[default] mode=custom application=/usr/local/bin/stream.sh format=pcm
allowing people from internet to connect to this asterisk instance
OR
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
; 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 AccessIt 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 phonesWe 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.
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