Difference between revisions of "Project:VOIP"

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A few members have been interested in doing something with VOIP/SIP things, mostly involving connecting the [[Phone|space phone line]] to the internet and doing voice menus, sip accounts for members, group chat, joining the [http://hackerspaces.org/wiki/Call-in Hackerspaces Call in], getting linked up to eventphone so we have a permanent extension and so on.
+
[[File:Gate_phone_1.jpg|400px|thumb|right|Our most frequently used extension - the back gate visitor phone]]
 +
[[File:Lhs-rshack-phone.png|400px|thumb|right| Cisco 7970 - the most common IP phone in the space]]
  
−
[[User:Fridgehead|Tom]] recently got hold of a pair of linksys/cisco SPA942 phones which are currently without power adapters. Will look for alternatives nearby.  ciborg_ is looking into getting some Cisco 7940/7970 phones donated.
+
A few members had been interested in playing with VOIP/SIP things, mostly involving connecting the [[Phone|space phone line]] to the internet and doing voice menus, sip accounts for members, group chat, joining the [http://hackerspaces.org/wiki/Call-in Hackerspaces Call in], getting linked up to other hackspaces, etc.  
  
−
----
+
==Functional features:==
 +
* Internal dialing between floors  (really easy rather than going back and forth back and forth to talk with someone / do something)
 +
* External calls get prompts for general and travel information.
 +
* <s>Doorbell-style ringtone on phone rings when back-gate visitors call  ('doorbell ringer' zombie-phone is located above Electronics lab)</s>
 +
* <s>Conference room function (mainly extension 4225)</s>
 +
* Weather-Resistant Outdoor Phones - [https://hubbellcdn.com/installationmanuals/SKU-GTC-Manual-502-20-0115-001.pdf GAI-Tronics Titan for Rail and Commander User and Installation Guide]
 +
* <s>''Back Door phone now with [[Robonaut|Robonaut]] integration - IRC users informed when a delivery call is made and also when not answered.''</s>
 +
* Toll-Free Outgoing dialing (0800 style numbers) / Banned dialing for others
 +
* DUE TO DUMB CALLERS DIALING FOR LENNY WHEN THEY ABSOLULUTELY SHOULDN'T HAVE AND STILL COMPLAINING ABOUT IT, WE HAD TO REMOVE THE [https://www.independent.co.uk/life-style/gadgets-and-tech/news/lenny-telemarketer-bot-robot-prank-a6813081.html LENNY FUNCTIONALITY.] RIP LENNY.
 +
* Easy networking to other Hackspaces
 +
* Dialing of Emergency numbers (101/111/112/999) (Please be aware that in the event of a power cut, hardware issue etc the phones will not work. Please use a mobile, or the payphone situated on High Road)
  
−
Functional features:
+
==PBX System details:==
−
  '''* System built on a tiny low-power RaspberryPi computer - minimising operating costs'''
 
−
  '''* Internal dialing between floors  (really easy rather than going back and forth back and forth to talk with someone / do something)'''
 
−
  '''* External incoming calls'''
 
−
  '''* Conference room function'''
 
−
  '''* Delivery gate functionality'''
 
−
  '''* Toll-Free Outgoing dialing (0800 style nums) / Banned dialing for others (recording: 'why not borrow a friend's mobile?) '''
 
−
  '''* General telephone information provided (address, nearest tube/train stops, membership information pointing to the website)'''
 
−
  * Paging / Intercom functionality (need to experiment with the actual physical phones to implement this)
 
−
  * Semi-secret comedy extensions (I'll document them in the wiki alongside the other stuff)
 
−
  * Easy networking to other Hackspaces and other projects (who entered the door last? Do we want a fortified intercom doorbell?)
 
−
<strike>* Dialing of Emergency 999 / 112 functionality <span style="color: red;> Decided to not be a liability with this (for now) </span> *** (outgoing call can be recorded and email alert sent to trustees and/or mailing list for safety) </strike>
 
  
 +
* System Platform: Hosted on [[Bell]], a managed Virtual Machine running on [[Equipment/Landin|Landin]]
 +
* Inbound and Outbound Connectivity to the PSTN is provided by [https://www.aaisp.net.uk/ Andrews & Arnold]
 +
* Internal phones are Cisco 79XX SIP handsets.
 +
* Outside phones are GAI-Tronics ruggedised telephones, both attached to a single Cisco ATA186 being powered over PoE using an Avaya 1603 700415607 PoE power splitter/injector.
 +
* OS: Debian Linux with tftpd-hpa service enabled for phones that need to boot and load config files over tftp, LHS logo screen graphics, etc.
 +
* PHONE SW: Asterisk  13.14.1 - no longer needing DAHDI as we've stopped using the TDM400 Digium FXO ports.
 +
* EXTERNAL VOICE GENERATION: Voice prompt files are currently manually generated using AWS Polly text to speech API.
 +
* CONFIG NOTE: Custom configs/scripts to be shared confidentially as they contain passwords.
  
−
(Above features in bold text have already been tested and enabled on the test PABX system)
+
==Asterisk Configuration==
  
−
Proposed extensions/ locations (specific locations to be appropriately and tastefully determined:
+
The asterisk configuration files can be found in <code>/etc/asterisk</code>. To make reading / managing the files easier they've been split up, and are "included" from the base config files.
−
  * Basement  - using Linksys SPA-942 for PC port passthrough in case someone needs to network a PC.
 
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  * Main floor - using Linksys SPA-941
 
−
  * Radio Shack - using Polycom conference unit or a Cisco 7940 phone
 
−
  * Outside gate - using Paul2's donated vandal-proof railroad analogue telephone handset feeding into the Cisco SPA122
 
−
  * Virtual extensions are OK, but we need to handle that accordingly.  
 
  
−
PABX System details:
+
===PJSIP===
−
  * HW: Hosted on one of the infrastructure VMs in the hack space (Was RaspberryPi Model B or B+ )
 
−
  * OS: Linux with tftpd service enabled for phones that need to boot and load config files over tftp, LHS logo screen graphics, etc.
 
−
  * PHONE SW: Asterisk software version 10 or 11 (Not truly necessary - we need something modern for ConfBridge)
 
−
  * EXTERNAL VOICE GENERATION: Voice prompt aiff/ulaw files generated from bash script on an OS X 10.9+ Mac using the "Kate" voice.
 
−
  * CONFIG NOTE: Custom configs/scripts shall be shared via github or posted on the wiki.
 
  
−
Infrastructure needs:
+
The SIP stack's core functionality, as well as endpoint definitions for endpoints such as phones and VoIP providers are managed by the chan_pjsip module. The syntax is somewhat different to Asterisks former chan_sip module, so ensure you're using the correct documentation when making changes.
−
  * Run ethernet to Radio Shack (and possibly biohacking lab?)
 
  
−
[[User:Kraptv|kraptv]] comments about some possible concerns about disturbing the peace:
+
{| class="wikitable"
−
''
+
! Filename
−
If people are annoyed at the phones nearby - they can press the "Do Not Disturb" button and it won't ring. In this design - I am mindful to keep a productive peace - I helped implement the phone system at a large cartoon company and was mindful to the variety of edge cases and balancing of sensitive users and collaborative users.
+
! Description
 +
|-
 +
| pjsip.conf
 +
| The default pjsip config template that ships with Asterisk. You shouldn't need to edit this file too much.
 +
|-
 +
| pjsip_transports.conf
 +
| Defines the various "transports", ie the IPs, ports and protocols Asterisk listens on.
 +
|-
 +
| pjsip_peers.conf
 +
| Defines "peers", E.G VoIP providers and other Hackspace's phone systems.
 +
|-
 +
| pjsip_extensions.conf
 +
| Defines "extensions", I.E Phones registering to the PBX
 +
|}
  
−
The best part of this: I've already built the majority of this on my home RaspberryPi with an 0845 number (I got it for free from sipgate.co.uk) so if people are interested in providing feedback on how the system works, I would love it.  Word of warning, if you're expecting to actually ring something at 447 Hackney Road, it won't be 'online' for maybe a month or so, but please test and provide suggestions. There's the chance I will answer the extension if I notice my soft-phone ringing.
+
===Extensions===
  
−
If it is completely crap and everyone hates it, we can scrap it easily and say 'man, corded phones are so 90's! Good riddance!' - but there are still many valid uses and an example of fun infrastructure enhancement. Noone is obligated to answer the phone. Noone has to touch it.  It's a co-operative, not some sort of business that has a telephone response service level agreement. ;-)''
+
Call routing and features are configured in the extensions files.
  
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Dial '''0845 594 7597''' to test the LHS phone system, please send feedback directly via email to [[User:Kraptv|kraptv]] or leave a voicemail!
+
{| class="wikitable"
 +
! Filename
 +
! Description
 +
|-
 +
| extensions.conf
 +
| The default extensions.conf template that ships with Asterisk. You shouldn't need to edit this file too much.
 +
|-
 +
| extensions_globals.conf
 +
| Defines various global variables such as ring groups, to make editing them easier.
 +
|-
 +
| extensions_internal.conf
 +
| Internal calls, i.e those originating from the various phones and ATAs in the Hackspace end up here.
 +
|-
 +
| extensions_external.conf
 +
| External calls, i.e those originating from the PSTN end up here.
 +
|-
 +
| extensions_barrier.conf
 +
| A temporary, experimental context, to facilitate opening the barrier from the phones.
 +
|-
 +
| extensions_utils.conf
 +
| Various utilities and functions are defined here.
 +
|}
  
−
* http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa_wip_admin.pdf
+
===Dial Plan Considerations===
−
* http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpipp/ip_phones/user/guide/spa9XX_user_web.pdf
+
VoIP phones have a configurable "Dial Plan" that decides when a call should be sent to the server. A good dial plan improves the user experience, as it means they don't have to press "dial" or wait for a timeout to kick in. The dial plan should also handle users prefixing calls with a 9, as this has become a (bad) habit for some.
 +
 
 +
The table below lays out the numbering plan used in the Hackspace. Emergency numbers should ''always'' go straight out, and the dialplan should be ordered so the user has to wait the minimum amount of time before the call is placed, eg an 11 digit number starting with a 0 should go straight out.
 +
 
 +
{| class="wikitable"
 +
! Number
 +
! Description
 +
! Notes
 +
|-
 +
| 999
 +
| Emergency Services
 +
|
 +
|-
 +
| 911
 +
| US Emergency Services
 +
| Should be re-written as 999
 +
|-
 +
| 112
 +
| Emergency Services (EU)
 +
|
 +
|-
 +
| 101
 +
| Non Emergency Police
 +
|
 +
|-
 +
| 105
 +
| Local Electricity Network Operator
 +
|
 +
|-
 +
| 111
 +
| Non Emergency Medical
 +
|
 +
|-
 +
| 888
 +
| Routed the same as 999, for testing.
 +
|
 +
|-
 +
| 100
 +
| Automated Operator
 +
|
 +
|-
 +
| 152
 +
| Echo Test
 +
|
 +
|-
 +
| 155
 +
| Hello World
 +
|
 +
|-
 +
| 1XX
 +
| Other Network Numbers
 +
|
 +
|-
 +
| 116XXX
 +
| European Helpline Numbers
 +
|
 +
|-
 +
| 6XXX
 +
| Internal Extensions
 +
|
 +
|-
 +
| 7XXX
 +
| Internal Short Codes for Ring Groups etc
 +
|
 +
|-
 +
| 8XXX
 +
| Reserved for DECT users
 +
|
 +
|-
 +
| 0ZXXXXXXXXX
 +
| UK Numbers
 +
| Blocked
 +
|-
 +
| 0ZX.
 +
| Shorter UK Numbers
 +
| Blocked
 +
|-
 +
| 00Z.
 +
| International Calls
 +
| Blocked
 +
|}
 +
 
 +
====Example Dialplan====
 +
 
 +
The following is an example Dialplan for a Linksys / Sipura / Newer Cisco device. Make sure if copying and pasting its a single line, with no whitespace.
 +
 
 +
'''!!! Currently Untested !!!'''
 +
 
 +
    (999S0|<9:>911S0|<9:>888S0|<9:>11[1234579]S0|<9:>116xxxS0|<9:>10xS0|<9:>15[0234568]S0|<9:>1x7[1-9]S0|<9:>6xxxS0|<9:>7xxxS0|<9:>8xxxS0|<9:>0[1-9]xxxxxxxxxS0|<9:>0[1-9]x.|<9:>00[1-9]x.|<9:>x.)
 +
 
 +
 
 +
==Infrastructure needs/Future uses:==
 +
* Mini-XML applications for the phones (weather, IRC info, relevant member info)
 +
* Enable emergency all-page public address functionality
 +
* [https://groups.google.com/forum/#!forum/interhackspace-pbx-project Connect with other Hackspaces]
 +
 
 +
Please use it, play with it, and hack on it to make it better!
 +
 
 +
==References:==
 +
 
 +
* http://www.asterisk.org/
 +
* http://www.digium.com/
 +
* http://www.gai-tronics.org.uk
 +
* http://www.cisco.com/

Latest revision as of 21:20, 15 December 2022

Our most frequently used extension - the back gate visitor phone
Cisco 7970 - the most common IP phone in the space

A few members had been interested in playing with VOIP/SIP things, mostly involving connecting the space phone line to the internet and doing voice menus, sip accounts for members, group chat, joining the Hackerspaces Call in, getting linked up to other hackspaces, etc.

Functional features:

  • Internal dialing between floors (really easy rather than going back and forth back and forth to talk with someone / do something)
  • External calls get prompts for general and travel information.
  • Doorbell-style ringtone on phone rings when back-gate visitors call ('doorbell ringer' zombie-phone is located above Electronics lab)
  • Conference room function (mainly extension 4225)
  • Weather-Resistant Outdoor Phones - GAI-Tronics Titan for Rail and Commander User and Installation Guide
  • Back Door phone now with Robonaut integration - IRC users informed when a delivery call is made and also when not answered.
  • Toll-Free Outgoing dialing (0800 style numbers) / Banned dialing for others
  • DUE TO DUMB CALLERS DIALING FOR LENNY WHEN THEY ABSOLULUTELY SHOULDN'T HAVE AND STILL COMPLAINING ABOUT IT, WE HAD TO REMOVE THE LENNY FUNCTIONALITY. RIP LENNY.
  • Easy networking to other Hackspaces
  • Dialing of Emergency numbers (101/111/112/999) (Please be aware that in the event of a power cut, hardware issue etc the phones will not work. Please use a mobile, or the payphone situated on High Road)

PBX System details:

  • System Platform: Hosted on Bell, a managed Virtual Machine running on Landin
  • Inbound and Outbound Connectivity to the PSTN is provided by Andrews & Arnold
  • Internal phones are Cisco 79XX SIP handsets.
  • Outside phones are GAI-Tronics ruggedised telephones, both attached to a single Cisco ATA186 being powered over PoE using an Avaya 1603 700415607 PoE power splitter/injector.
  • OS: Debian Linux with tftpd-hpa service enabled for phones that need to boot and load config files over tftp, LHS logo screen graphics, etc.
  • PHONE SW: Asterisk 13.14.1 - no longer needing DAHDI as we've stopped using the TDM400 Digium FXO ports.
  • EXTERNAL VOICE GENERATION: Voice prompt files are currently manually generated using AWS Polly text to speech API.
  • CONFIG NOTE: Custom configs/scripts to be shared confidentially as they contain passwords.

Asterisk Configuration

The asterisk configuration files can be found in /etc/asterisk. To make reading / managing the files easier they've been split up, and are "included" from the base config files.

PJSIP

The SIP stack's core functionality, as well as endpoint definitions for endpoints such as phones and VoIP providers are managed by the chan_pjsip module. The syntax is somewhat different to Asterisks former chan_sip module, so ensure you're using the correct documentation when making changes.

Filename Description
pjsip.conf The default pjsip config template that ships with Asterisk. You shouldn't need to edit this file too much.
pjsip_transports.conf Defines the various "transports", ie the IPs, ports and protocols Asterisk listens on.
pjsip_peers.conf Defines "peers", E.G VoIP providers and other Hackspace's phone systems.
pjsip_extensions.conf Defines "extensions", I.E Phones registering to the PBX

Extensions

Call routing and features are configured in the extensions files.

Filename Description
extensions.conf The default extensions.conf template that ships with Asterisk. You shouldn't need to edit this file too much.
extensions_globals.conf Defines various global variables such as ring groups, to make editing them easier.
extensions_internal.conf Internal calls, i.e those originating from the various phones and ATAs in the Hackspace end up here.
extensions_external.conf External calls, i.e those originating from the PSTN end up here.
extensions_barrier.conf A temporary, experimental context, to facilitate opening the barrier from the phones.
extensions_utils.conf Various utilities and functions are defined here.

Dial Plan Considerations

VoIP phones have a configurable "Dial Plan" that decides when a call should be sent to the server. A good dial plan improves the user experience, as it means they don't have to press "dial" or wait for a timeout to kick in. The dial plan should also handle users prefixing calls with a 9, as this has become a (bad) habit for some.

The table below lays out the numbering plan used in the Hackspace. Emergency numbers should always go straight out, and the dialplan should be ordered so the user has to wait the minimum amount of time before the call is placed, eg an 11 digit number starting with a 0 should go straight out.

Number Description Notes
999 Emergency Services
911 US Emergency Services Should be re-written as 999
112 Emergency Services (EU)
101 Non Emergency Police
105 Local Electricity Network Operator
111 Non Emergency Medical
888 Routed the same as 999, for testing.
100 Automated Operator
152 Echo Test
155 Hello World
1XX Other Network Numbers
116XXX European Helpline Numbers
6XXX Internal Extensions
7XXX Internal Short Codes for Ring Groups etc
8XXX Reserved for DECT users
0ZXXXXXXXXX UK Numbers Blocked
0ZX. Shorter UK Numbers Blocked
00Z. International Calls Blocked

Example Dialplan

The following is an example Dialplan for a Linksys / Sipura / Newer Cisco device. Make sure if copying and pasting its a single line, with no whitespace.

!!! Currently Untested !!! 

   (999S0|<9:>911S0|<9:>888S0|<9:>11[1234579]S0|<9:>116xxxS0|<9:>10xS0|<9:>15[0234568]S0|<9:>1x7[1-9]S0|<9:>6xxxS0|<9:>7xxxS0|<9:>8xxxS0|<9:>0[1-9]xxxxxxxxxS0|<9:>0[1-9]x.|<9:>00[1-9]x.|<9:>x.)


Infrastructure needs/Future uses:

  • Mini-XML applications for the phones (weather, IRC info, relevant member info)
  • Enable emergency all-page public address functionality
  • Connect with other Hackspaces

Please use it, play with it, and hack on it to make it better!

References:

Retrieved from ‘https://wiki.london.hackspace.org.uk/w/index.php?title=Project:VOIP&oldid=55383’