SPA-962

VoIP Hardware Solutions
Provider Solution Details
VoIP Hardware Zycoo UC Solutions
  • Modular Design IP PBX for SMB
  • Remote office Centralized Management solution
  • 3rd party app integration, Enterprise Billing, Android & iOS client
Details
Yeastar Communications Solutions
  • Cost-effective IP-PBX Solution for SMB
  • FXS, FXO, GSM, BRI and PRI VoIP Gateways
  • Rich features and reliable performance
Details
Information from CeBit indicates the following:
  • Six Line Business Telephone Based on the SIP Standard
  • High Resolution Color Display
  • Two 10/100 Ethernet Ports with Integrated Ethernet Switch
  • Supports 802.3af
  • Power Over Ethernet PoE or External Power Adapter

Photos:
http://flickr.com/photos/voipnovatos/sets/72157594376507389/

Reviews

Linksys SPA962 Review

Support





Product Description from Linksys' website:


Stylish and functional in design, the SPA962 VoIP telephone is a must for businesses using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA962 leverages industry leading VoIP technology from Linksys to deliver a high quality IP Phone that is unparalleled in features, value, and support.

Standard features on the SPA962 include six active lines, dual switched Ethernet ports, 802.3af PoE support, a high resolution color display, speakerphone, and a 2.5 mm head-set port. Each line can be independently configured to use a unique phone number (or extension), or can be configured to use a shared number that is assigned to multiple phones. The power supply for the SPA962 is sold separately and will be required if PoE functionality is not implemented.

Comprehensive Interoperability and SIP Based Feature Set Based on the SIP standard, the SPA962 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable service parameters, the SPA962 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA962.

Carrier-Grade Security, Provisioning, and Management The SPA962 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading, and re-configuring customer premise equipment (CPE).


From the Data Sheet on Linksys' website:

SPA962 Key Telephone Functions and Features

� Up to Six Lines with Independent Configuration and Registration
� 320 x 240 True Color, Four Inch, Liquid Crystal Display (LCD)
� Secure Call Support - SIP over TLS, and SRTP
� Line Status - Active Line Indication, Name and Number
� Menu Driven User Interface - Multiple Languages Supported
� Digits Dialed with Number Auto-Completion
� Shared / Bridged Line Appearance **
� High Quality Speakerphone
� Call Hold
� Music on Hold **
� Call Waiting
� Caller ID Name and Number and Outbound Caller ID Blocking
� Outbound Caller ID Blocking
� Call Transfer - Attended and Blind
� Call Conferencing
� Automatic Redial
� On-Hook Dialing
� Call Pick Up - Selective and Group **
� Call Park and UnPark **
� Call Swap
� Call Back on Busy
� Call Blocking - Anonymous and Selective
� Call Forwarding - Unconditional, No Answer, On Busy
� Hot Line and Warm Line Automatic Calling
� Call Logs (60 entries each): Made, Answered, and Missed Calls
� Redial from Call Logs
� Personal Directory with Auto-dial (100 entries)
� Do Not Disturb (callers hear line busy tone)
� URI (IP) Dialing Support (Vanity Numbers)
� On Hook Default Audio Configuration (Speakerphone and Headset)
� Multiple Ring Tones with Selectable Ring Tone per Line
� Called Number with Directory Name Matching
� Call Number using Name - Directory Matching or via Caller ID
� Subsequent Incoming Calls with Calling Name and Number
� Date and Time with Intelligent Daylight Savings Support
� Call Duration and Start Time Stored in Call Logs
� Call Timer
� Name and Identity (Text) Displayed at Start Up
� Distinctive Ringing Based on Calling and Called Number
� Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com
� Speed Dialing
� Configurable Dial/Numbering Plan Support - per Line
� Intercom **
� Group Paging **
� DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
� Syslog, Debug, Report Generation, and Event Logging
� Secure Call Encrypted Voice Communication Support - SIP over TLS, and SRTP
� Built-in Web Server for Administration and Configuration with Multiple Security Levels
� Automated Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
� Optionally Require Admin Password to Reset Unit to factory Defaults
    • Feature requires support by SIP server

Hardware
� 320 x 240 True Color, Four Inch, Liquid Crystal Display (LCD)
� Four Illuminated Call Appearance Line Buttons with Tricolor LED's
� LED Indicates Line State � Active, Idle, On-Hold, Unregistered
� Line LED Configurable to 13 Different States (On/Off, Color, Flash)
� Dedicated Illuminated Buttons for:
� Audio Mute On/Off
� Headset On/Off
� Speakerphone On/Off
� Four Soft Key Buttons
� Four Way Rocking Directional Knob for Menu Navigation
� Support for up to two attached Attendant Consoles; adds up to 64 programmable buttons
� Voice Mail Message Waiting Indicator Light
� Voice Mail Message Retrieval Button
� Dedicated Hold Button
� Settings Button for Access to Feature, Set-up, and Configuration Menus
� Volume Control Rocking Up/Down Knob Controls Handset, Headset, Speaker, Ringer
� Standard 12-Button Dialing Pad
� High Quality Handset and Cradle
� Built-In High Quality Microphone and Speaker
� Headset Jack � 2.5 millimeter
� LED Test Function
� Two Ethernet LAN Ports with Integrated Ethernet Switch � 100BaseT RJ-45
� 802.3af Compliant Power over Ethernet (PoE)
� Optional 5 volt DC Universal (100-240 Volt) Switching - Power Supply is Ordered Separately

SPA-932 Sidecar Busy Lamp Fields + Speed Dials


To get the SPA 932 sidecar working with your SPA 962 phone and Asterisk, you need to do the following:

  1. Download the lastest firmware for the phone from the Linksys website, and upgrade the phone
  2. Find the IP address of the phone
    1. Press the key on the phone that looks like a piece of paper with the corner folded. (This is often called the "Settings" or "Menu" key)
    2. Select the option that says "Network". (This is option number 9, and you can jump directly to this option by pressing the 9 key)
    3. The second item on that page called CurrentIP is the current IP address of that phone
  3. Go to the web interface of the phone, and get into advanced admin mode
    1. In a web browser, type http://a.b.c.d/ in the address bar, where a.b.c.d is the IP address you found in the previous step
    2. In the upper-right-hand corner of the web interface, select "admin". As soon as the page reloads, select "advanced".
  4. Configure the SPA-932 for Asterisk
    1. In the web interface, look for the tabs at the top of the page. Click the tab that says "SPA932" (it should be the very last tab)
    2. Find the drop-down box labeled "Server Type" and set it to "Asterisk"
    3. Under the heading "Unit 1", you'll see a blank line for each of the buttons on the SPA 932 sidecar. Each of these can be configured for a different BLF (busy lamp field) and speed dial
    4. The syntax for the line is "fnc=blf+sd;sub=201@192.168.16.105". The "blf+sd" tells the phone to make this button both a BLF and a speed dial. The 201 tells the phone to subscribe to the state of extension 201 at the Asterisk server, which in this case has an IP address of 192.168.16.105.
  5. Configure Asterisk for presence information
    1. Add hints to the dialplan as appropriate (see Asterisk Presence)
    2. Set call limits on your SIP peers (Asterisk won't keep track of device state on SIP devices unless a call limit is enforced)
    3. Make sure the "subscribecontext" setting is set to the correct dialplan context

I've placed snippets of my configs here below:

On the SPA932 tab on the phone:
Server type: Asterisk
Unit 1 Key 1: fnc=blf+sd;sub=201@192.168.16.105
Unit 1 Key 2: fnc=blf+sd;sub=202@192.168.16.105
Unit 1 Key 3: fnc=blf+sd;sub=203@192.168.16.105
Unit 1 Key 4: fnc=sd;sub=98005551212;nme=AT&T Directory Assist [exception: this is a speed dial only!]

In sip.conf on the Asterisk server:
[general]
subscribecontext=lab

[linksys1]
type=friend
secret=linksys
context=lab
disallow=all
allow=ulaw
allow=gsm
qualify=no
host=dynamic
call-limit=99

[linksys2]
type=friend
secret=linksys
context=lab
disallow=all
allow=ulaw
allow=gsm
qualify=no
host=dynamic
call-limit=99

[linksys3]
type=friend
secret=linksys
context=lab
disallow=all
allow=ulaw
allow=gsm
qualify=no
host=dynamic
call-limit=99

In extensions.conf, the dialplan on the Asterisk server:
[lab]
exten => 201,1,Dial(SIP/linksys1)
exten => 201,hint,SIP/linksys1

exten => 202,1,Dial(SIP/linksys2)
exten => 202,hint,SIP/linksys2

exten => 203,1,Dial(SIP/linksys3)
exten => 203,hint,SIP/linksys3

SPA-932 Pickup


Pickup with this phone or firmware doesn't work, may be only with SPA900. BLF works OK but when a phone is ringing and you press its button, SPA962 just dials the extension number. We managed to make it work with a little trick:

Add this to extensions.conf (using freepbx for example in file extensions_custom.conf context "from-internal-custom") :

exten => _*00XXX,1,Dial(SIP/${EXTEN:3},60,Tt)
exten => _*00XXX,2,Pickupchan(SIP/${EXTEN:3})
exten => *00201,hint,SIP/201
exten => *00202,hint,SIP/202
exten => *00203,hint,SIP/203
exten => *00204,hint,SIP/204
exten => *00205,hint,SIP/205
exten => *00206,hint,SIP/206
...


You must change 2XX with your extension numbers. And yes, you cannot use wildcards on hint extensions.

Finally on SPA932 set button configuration as this:

fnc=sd+blf+cp;sub=*00201@A.B.C.D;


Where A.B.C.D is the IP address of your PBX


This way in Asterisk we have associated hint of *002XX with the presence of extension 2XX and if you press this button and dial *00208:
- if extension is idle it will ring
- if it is ringing you will pick up it.


Alternative Pickup


With the release of the later firmwares, you can now enable call pickups on the SPA932 sidecar with Asterisk's default call pickup code of **.
Configure the Call Pickup Code in the SPA932 panel as **#
Configure your buttons as usual:
fnc=blf+sd+cp;sub=101@10.10.10.1;ext=101;nme=Reception

Pressing the button when the device is in a ringing state will now dial **#, replacing # with the 'ext' number.


Tutorials:




Where to Buy:


Information from CeBit indicates the following:
  • Six Line Business Telephone Based on the SIP Standard
  • High Resolution Color Display
  • Two 10/100 Ethernet Ports with Integrated Ethernet Switch
  • Supports 802.3af
  • Power Over Ethernet PoE or External Power Adapter

Photos:
http://flickr.com/photos/voipnovatos/sets/72157594376507389/

Reviews

Linksys SPA962 Review

Support





Product Description from Linksys' website:


Stylish and functional in design, the SPA962 VoIP telephone is a must for businesses using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA962 leverages industry leading VoIP technology from Linksys to deliver a high quality IP Phone that is unparalleled in features, value, and support.

Standard features on the SPA962 include six active lines, dual switched Ethernet ports, 802.3af PoE support, a high resolution color display, speakerphone, and a 2.5 mm head-set port. Each line can be independently configured to use a unique phone number (or extension), or can be configured to use a shared number that is assigned to multiple phones. The power supply for the SPA962 is sold separately and will be required if PoE functionality is not implemented.

Comprehensive Interoperability and SIP Based Feature Set Based on the SIP standard, the SPA962 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable service parameters, the SPA962 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA962.

Carrier-Grade Security, Provisioning, and Management The SPA962 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading, and re-configuring customer premise equipment (CPE).


From the Data Sheet on Linksys' website:

SPA962 Key Telephone Functions and Features

� Up to Six Lines with Independent Configuration and Registration
� 320 x 240 True Color, Four Inch, Liquid Crystal Display (LCD)
� Secure Call Support - SIP over TLS, and SRTP
� Line Status - Active Line Indication, Name and Number
� Menu Driven User Interface - Multiple Languages Supported
� Digits Dialed with Number Auto-Completion
� Shared / Bridged Line Appearance **
� High Quality Speakerphone
� Call Hold
� Music on Hold **
� Call Waiting
� Caller ID Name and Number and Outbound Caller ID Blocking
� Outbound Caller ID Blocking
� Call Transfer - Attended and Blind
� Call Conferencing
� Automatic Redial
� On-Hook Dialing
� Call Pick Up - Selective and Group **
� Call Park and UnPark **
� Call Swap
� Call Back on Busy
� Call Blocking - Anonymous and Selective
� Call Forwarding - Unconditional, No Answer, On Busy
� Hot Line and Warm Line Automatic Calling
� Call Logs (60 entries each): Made, Answered, and Missed Calls
� Redial from Call Logs
� Personal Directory with Auto-dial (100 entries)
� Do Not Disturb (callers hear line busy tone)
� URI (IP) Dialing Support (Vanity Numbers)
� On Hook Default Audio Configuration (Speakerphone and Headset)
� Multiple Ring Tones with Selectable Ring Tone per Line
� Called Number with Directory Name Matching
� Call Number using Name - Directory Matching or via Caller ID
� Subsequent Incoming Calls with Calling Name and Number
� Date and Time with Intelligent Daylight Savings Support
� Call Duration and Start Time Stored in Call Logs
� Call Timer
� Name and Identity (Text) Displayed at Start Up
� Distinctive Ringing Based on Calling and Called Number
� Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com
� Speed Dialing
� Configurable Dial/Numbering Plan Support - per Line
� Intercom **
� Group Paging **
� DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
� Syslog, Debug, Report Generation, and Event Logging
� Secure Call Encrypted Voice Communication Support - SIP over TLS, and SRTP
� Built-in Web Server for Administration and Configuration with Multiple Security Levels
� Automated Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
� Optionally Require Admin Password to Reset Unit to factory Defaults
    • Feature requires support by SIP server

Hardware
� 320 x 240 True Color, Four Inch, Liquid Crystal Display (LCD)
� Four Illuminated Call Appearance Line Buttons with Tricolor LED's
� LED Indicates Line State � Active, Idle, On-Hold, Unregistered
� Line LED Configurable to 13 Different States (On/Off, Color, Flash)
� Dedicated Illuminated Buttons for:
� Audio Mute On/Off
� Headset On/Off
� Speakerphone On/Off
� Four Soft Key Buttons
� Four Way Rocking Directional Knob for Menu Navigation
� Support for up to two attached Attendant Consoles; adds up to 64 programmable buttons
� Voice Mail Message Waiting Indicator Light
� Voice Mail Message Retrieval Button
� Dedicated Hold Button
� Settings Button for Access to Feature, Set-up, and Configuration Menus
� Volume Control Rocking Up/Down Knob Controls Handset, Headset, Speaker, Ringer
� Standard 12-Button Dialing Pad
� High Quality Handset and Cradle
� Built-In High Quality Microphone and Speaker
� Headset Jack � 2.5 millimeter
� LED Test Function
� Two Ethernet LAN Ports with Integrated Ethernet Switch � 100BaseT RJ-45
� 802.3af Compliant Power over Ethernet (PoE)
� Optional 5 volt DC Universal (100-240 Volt) Switching - Power Supply is Ordered Separately

SPA-932 Sidecar Busy Lamp Fields + Speed Dials


To get the SPA 932 sidecar working with your SPA 962 phone and Asterisk, you need to do the following:

  1. Download the lastest firmware for the phone from the Linksys website, and upgrade the phone
  2. Find the IP address of the phone
    1. Press the key on the phone that looks like a piece of paper with the corner folded. (This is often called the "Settings" or "Menu" key)
    2. Select the option that says "Network". (This is option number 9, and you can jump directly to this option by pressing the 9 key)
    3. The second item on that page called CurrentIP is the current IP address of that phone
  3. Go to the web interface of the phone, and get into advanced admin mode
    1. In a web browser, type http://a.b.c.d/ in the address bar, where a.b.c.d is the IP address you found in the previous step
    2. In the upper-right-hand corner of the web interface, select "admin". As soon as the page reloads, select "advanced".
  4. Configure the SPA-932 for Asterisk
    1. In the web interface, look for the tabs at the top of the page. Click the tab that says "SPA932" (it should be the very last tab)
    2. Find the drop-down box labeled "Server Type" and set it to "Asterisk"
    3. Under the heading "Unit 1", you'll see a blank line for each of the buttons on the SPA 932 sidecar. Each of these can be configured for a different BLF (busy lamp field) and speed dial
    4. The syntax for the line is "fnc=blf+sd;sub=201@192.168.16.105". The "blf+sd" tells the phone to make this button both a BLF and a speed dial. The 201 tells the phone to subscribe to the state of extension 201 at the Asterisk server, which in this case has an IP address of 192.168.16.105.
  5. Configure Asterisk for presence information
    1. Add hints to the dialplan as appropriate (see Asterisk Presence)
    2. Set call limits on your SIP peers (Asterisk won't keep track of device state on SIP devices unless a call limit is enforced)
    3. Make sure the "subscribecontext" setting is set to the correct dialplan context

I've placed snippets of my configs here below:

On the SPA932 tab on the phone:
Server type: Asterisk
Unit 1 Key 1: fnc=blf+sd;sub=201@192.168.16.105
Unit 1 Key 2: fnc=blf+sd;sub=202@192.168.16.105
Unit 1 Key 3: fnc=blf+sd;sub=203@192.168.16.105
Unit 1 Key 4: fnc=sd;sub=98005551212;nme=AT&T Directory Assist [exception: this is a speed dial only!]

In sip.conf on the Asterisk server:
[general]
subscribecontext=lab

[linksys1]
type=friend
secret=linksys
context=lab
disallow=all
allow=ulaw
allow=gsm
qualify=no
host=dynamic
call-limit=99

[linksys2]
type=friend
secret=linksys
context=lab
disallow=all
allow=ulaw
allow=gsm
qualify=no
host=dynamic
call-limit=99

[linksys3]
type=friend
secret=linksys
context=lab
disallow=all
allow=ulaw
allow=gsm
qualify=no
host=dynamic
call-limit=99

In extensions.conf, the dialplan on the Asterisk server:
[lab]
exten => 201,1,Dial(SIP/linksys1)
exten => 201,hint,SIP/linksys1

exten => 202,1,Dial(SIP/linksys2)
exten => 202,hint,SIP/linksys2

exten => 203,1,Dial(SIP/linksys3)
exten => 203,hint,SIP/linksys3

SPA-932 Pickup


Pickup with this phone or firmware doesn't work, may be only with SPA900. BLF works OK but when a phone is ringing and you press its button, SPA962 just dials the extension number. We managed to make it work with a little trick:

Add this to extensions.conf (using freepbx for example in file extensions_custom.conf context "from-internal-custom") :

exten => _*00XXX,1,Dial(SIP/${EXTEN:3},60,Tt)
exten => _*00XXX,2,Pickupchan(SIP/${EXTEN:3})
exten => *00201,hint,SIP/201
exten => *00202,hint,SIP/202
exten => *00203,hint,SIP/203
exten => *00204,hint,SIP/204
exten => *00205,hint,SIP/205
exten => *00206,hint,SIP/206
...


You must change 2XX with your extension numbers. And yes, you cannot use wildcards on hint extensions.

Finally on SPA932 set button configuration as this:

fnc=sd+blf+cp;sub=*00201@A.B.C.D;


Where A.B.C.D is the IP address of your PBX


This way in Asterisk we have associated hint of *002XX with the presence of extension 2XX and if you press this button and dial *00208:
- if extension is idle it will ring
- if it is ringing you will pick up it.


Alternative Pickup


With the release of the later firmwares, you can now enable call pickups on the SPA932 sidecar with Asterisk's default call pickup code of **.
Configure the Call Pickup Code in the SPA932 panel as **#
Configure your buttons as usual:
fnc=blf+sd+cp;sub=101@10.10.10.1;ext=101;nme=Reception

Pressing the button when the device is in a ringing state will now dial **#, replacing # with the 'ext' number.


Tutorials:




Where to Buy:


Created by: jblachly, Last modification: Fri 28 of Jan, 2011 (15:03 UTC) by VoilensP
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