3.2.23. Linksys SPA942
Linksys SPA942
Feature List
Technical Specifications
Installation
Notes:
Here is a quick step by step installation of Linksys SPA942:
Accounts Configuration
Prerequisitives
Before going further reading this tutorial make sure you have asterisk server and you are familiar with adding users to asterisk. If you are not aware of this issue you can read the tutorial explaining how to add new users to asterisk.
Linksys SPA942 is working with the SIP (Session Initiation Protocol). You will have to create a SIP user. Open /etc/asterisk/sip.conf and add the user at the bottom at the file. Since the hardphone support more than one lines we will create two accounts for the hardphone
Then reload Asterisk's sip configuration by executing the command `asterisk -rx "sip reload"`.
At one point we will want to receive calls on our phone, that's why we have to create an extension for this. We will achieve this by adding extension `1001` in our default context in extensions.conf configuration file, located at /etc/asterisk. We will also create an extension `1002` which will dial the second account.
Of course we will have to reload our asterisk extension configuration in order the changes to take effect. We can easily achieve this by typing:
In order to use the voicemail features of the Linksys SPA942 we will create two mailbox accounts. In order to achieve this we will have to edit voicemail.conf. Here are the lines that we have to add:
You will have to make some modifications in the dialplan in order the voicemail to work properly. First well add Voicemail application in the part where we dial the two accounts.
Then we will create two extensions which purpose will be to check the mailboxes accounts. Here are the lines that we are going to add to extensions.conf
exten => 111001,1,Answer()
exten => 111001,2,VoiceMailMain(11001@default)
exten => 111001,n,HangUp()
exten => 111002,1,Answer()
exten => 111002,2,VoiceMailMain(11002@default)
exten => 111002,n,HangUp() We will have to reload the dialplan and the voicemail configuration in order the changes to take effect. We will achieve this by using the command
You can download the complete configuration files from our server:
Initialization
When you power up the phone it will start up. It will boot its operational system. When its ready you will be able to press the options button (the one which looks like page). In order to operate with the phone you will have to configure some options.
Network - network options are the the ninth option. Use the navigation buttons and press Select.
SIP Options - in order to configure the account on the phone you will have to use the web-interface. In order to have access to it you will need to configure you network settings and enable the web interface from the phone menus (it is enabled by default). Type the IP address in your browser and enter into administration mode by clicking `Admin Mode`. You will be prompted for password if you have set such. Then go to Ext 1 tab as is shown below.
This is the location where you can configure your account.
Repeat this operation with the second SIP account that we created (mytest2) but on Ext 2
Now you are ready to use your Linksys SPA942. In the next pages you will find some advances options and configurations of the hardphone.
Using the Web interface
Most of the options are accessible through the web configuration interface. In order to access the web interface you will need a correctly configured network. An example is shown on the previous page.
The web interface is divided into seven tabs. Every tab gives different information. Here are the eight tabs:
The first tab named `Info` will show you different information for your hardphone.
The second tab named `System` will allow you to reconfigure hardphone's network configuration.
The third tab named `SIP` will allow you the configure some advanced SIP options
The fourth tab named `Regional` will allow you to configure some Regional settings as well as the service activation and deactivation codes
From the fifth tab named `Phone` you will be able to set some settings concerning the user such as sounds, naming, logos, etc...
From the sixth and seventh tabs named `Ext 1` and `Ext 2`you will be able to set the SIP settings for Ext 1 and Ext 2
From the last tab named `User` you will be able to set additional user options such as Call Forwarding, Speed Dialing and Audio settings.
Links
Here we have gathered some useful links and resources:
Feature List
Up to Four (4) Lines with Independent Configuration/Registration. The SPA941 ships with two (2) line appearances enabled. A two (2) line upgrade is available via a software license key installed locally using the SPA941 web interface or remotely via a secure profile update.
Pixel Based Display: 128x64 Monochrome LCD Graphical Display
Line Status - Active Line Indication, Name/Number
Menu Driven User Interface
Three Party Call Conference Support
Digits Dialed with Number Auto-Completion. Speed Dial Support
Shared Line Appearance Support
Hands Free Operation
call hold, call waiting
Music on Hold Support
Caller ID Name and Number & Outbound Caller ID Blocking
Call Transfer - Attended and Blind
Automatic Redial
Call Forwarding - Unconditional, No Answer, On Busy
Call park, call pick up, call swap
Hot Line and Warm Line Automatic Calling
URI (IP) Dialing Support (Vanity Numbers)
Personal Directory with Auto-dial (100 entries)
Multiple Ring Tones with Selectable Default Ring Tone per Line. User Downloadable Ring Tones and Ring Tone Generator (Free from www.sipura.com)
Call Log (60 entries each): Made, Answered, Missed Calls
Call Blocking - Anonymous and Selective
On Hook Default Audio Configuration (Hands Free/Headset)
Call back on busy. Do Not Disturb
Configurable Dial/Numbering Plan Support - per Line
Built-in Web Server for Admin and Config with Multiple Security Levels
Called Number with Directory Name Matching. Calling Number with Name - Directory Matching or via Caller ID
SecureCall Encrypted Voice Communication Support
DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
Syslog, Debug, Report Generation and Event Logging
Automated Provisioning, Multiple Schemes-Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
Require Admin Password to Reset Unit to factory Defaults Option
Line Status - Active Line Indication, Name/Number
Menu Driven User Interface
Three Party Call Conference Support
Digits Dialed with Number Auto-Completion. Speed Dial Support
Shared Line Appearance Support
Hands Free Operation
call hold, call waiting
Music on Hold Support
Caller ID Name and Number & Outbound Caller ID Blocking
Call Transfer - Attended and Blind
Automatic Redial
Call Forwarding - Unconditional, No Answer, On Busy
Call park, call pick up, call swap
Hot Line and Warm Line Automatic Calling
URI (IP) Dialing Support (Vanity Numbers)
Personal Directory with Auto-dial (100 entries)
Multiple Ring Tones with Selectable Default Ring Tone per Line. User Downloadable Ring Tones and Ring Tone Generator (Free from www.sipura.com)
Call Log (60 entries each): Made, Answered, Missed Calls
Call Blocking - Anonymous and Selective
On Hook Default Audio Configuration (Hands Free/Headset)
Call back on busy. Do Not Disturb
Configurable Dial/Numbering Plan Support - per Line
Built-in Web Server for Admin and Config with Multiple Security Levels
Called Number with Directory Name Matching. Calling Number with Name - Directory Matching or via Caller ID
SecureCall Encrypted Voice Communication Support
DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
Syslog, Debug, Report Generation and Event Logging
Automated Provisioning, Multiple Schemes-Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
Require Admin Password to Reset Unit to factory Defaults Option
Technical Specifications
Hardware
Software
Power supply: DC Input Voltage: +5 VDC at 2.0 A Max. 100-240v -
50-60Hz (26-34VA) AC Input, 1.8m cord
Temp range: 41 to 113 Ί F (5 to 45 Ί C)
Humidity: 10 to 90% Non-condensing, operating and non-operating
Display: Pixel Based Display: 128x64 Monochrome LCD Graphical Display
Keyboard: Dedicated Illuminated Buttons for: Audio Mute On/Off, Headset On/Off, Speakerphone On/Off. Four (4) Soft Key Buttons. Four-Way Directional Button for GUI Navigation. Voice Mail Message Retrieval Button. Dedicated Hold Button. Settings Button for Access to Feature, Set-up & Configuration Menus. Volume Control Button Handset, Headset, Speaker, Ringer. Standard 12-Button Dialing Pad. Mute Button with LED.
Lights: Four (4) Illuminated Call Appearance/Line Buttons with Tricolor LED, Line LED State Indication Active, Idle, On Hold, Unregistered. Line LED Configurable to 13 Different States (On/Off, Color, Flash). Message Waiting Indicator Light. LED Test Function.
Audio: Built-in Speakerphone and Microphone.
Network: (1) 10/100 BaseT E auto-detect RJ45 (Ethernet), Handset: RJ7 Connector Standard
Headset: 2.5 mm
50-60Hz (26-34VA) AC Input, 1.8m cord
Temp range: 41 to 113 Ί F (5 to 45 Ί C)
Humidity: 10 to 90% Non-condensing, operating and non-operating
Display: Pixel Based Display: 128x64 Monochrome LCD Graphical Display
Keyboard: Dedicated Illuminated Buttons for: Audio Mute On/Off, Headset On/Off, Speakerphone On/Off. Four (4) Soft Key Buttons. Four-Way Directional Button for GUI Navigation. Voice Mail Message Retrieval Button. Dedicated Hold Button. Settings Button for Access to Feature, Set-up & Configuration Menus. Volume Control Button Handset, Headset, Speaker, Ringer. Standard 12-Button Dialing Pad. Mute Button with LED.
Lights: Four (4) Illuminated Call Appearance/Line Buttons with Tricolor LED, Line LED State Indication Active, Idle, On Hold, Unregistered. Line LED Configurable to 13 Different States (On/Off, Color, Flash). Message Waiting Indicator Light. LED Test Function.
Audio: Built-in Speakerphone and Microphone.
Network: (1) 10/100 BaseT E auto-detect RJ45 (Ethernet), Handset: RJ7 Connector Standard
Headset: 2.5 mm
Software
Configuration: Web Browser Administration & Configuration via Integral
Web Server. Telephone Key Pad Configuration via Display GUI Menu / Navigation. Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP. Asynchronous Notification of Upgrade Availability via NOTIFY.
Protocols Internet: IPv4, ARP, DNS, DHCP, ICMP, TCP, UDP, RTP, RTCP, Diffserv, VLAN, SNTP
VoIP: SIPv2, SIP Proxy Redundancy - Dynamic via DNS SRV, A Records, SIP Support in Network Address Translation Networks - NAT (incl. STUN), Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP. Jitter Buffer - Adaptive. Jitter Buffer - Adaptive. VAD - Voice Activity Detection w/ Silence Suppression .
Voice encoding: G.711 aLaw, uLaw, G.726 (16, 24, 32, 40Kbs), G.729A, G723.1 (6.3, 5.5 Kbps), DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO)
Security: Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP. Password Protected System Reset to Factory Default. Password Protected Admin and User Access Authority. HTTPS with Factory Installed Client Certificate. HTTP Digest - Encrypted Authentication via MD5 (RFC 1321). Up to 256-bit AES Encryption.
Web Server. Telephone Key Pad Configuration via Display GUI Menu / Navigation. Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP. Asynchronous Notification of Upgrade Availability via NOTIFY.
Protocols Internet: IPv4, ARP, DNS, DHCP, ICMP, TCP, UDP, RTP, RTCP, Diffserv, VLAN, SNTP
VoIP: SIPv2, SIP Proxy Redundancy - Dynamic via DNS SRV, A Records, SIP Support in Network Address Translation Networks - NAT (incl. STUN), Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP. Jitter Buffer - Adaptive. Jitter Buffer - Adaptive. VAD - Voice Activity Detection w/ Silence Suppression .
Voice encoding: G.711 aLaw, uLaw, G.726 (16, 24, 32, 40Kbs), G.729A, G723.1 (6.3, 5.5 Kbps), DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO)
Security: Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP. Password Protected System Reset to Factory Default. Password Protected Admin and User Access Authority. HTTPS with Factory Installed Client Certificate. HTTP Digest - Encrypted Authentication via MD5 (RFC 1321). Up to 256-bit AES Encryption.
Installation
Notes:
a. The Linksys SPA942 should be placed on horizontal surface.
b. Use the external power supply that is included in the package. Other power supplies may cause damage to the phone, affect the behavior or induce noise.
b. Use the external power supply that is included in the package. Other power supplies may cause damage to the phone, affect the behavior or induce noise.
Here is a quick step by step installation of Linksys SPA942:
1. Put the stand of the phone
2. Plug the one end of the handset's cable into the phone and the other into the handset itself.
3. Plug your Internet cable into the port with label 'WAN'.
4. Plug your PC cable into the port with label 'PC'. Plug the other end into the Ethernet port of your PC. In doing this you will ensure that both the phone and your PC will be supplied with Internet without need of third device or second Internet line.
5. Power up the phone by plugging the power supply into the phone.
6. If you have headset plug it into the port with 'headset' image near it.
7. Attaching the Desk Stand
Note: Note that points 4 and 6 are optional.
2. Plug the one end of the handset's cable into the phone and the other into the handset itself.
3. Plug your Internet cable into the port with label 'WAN'.
4. Plug your PC cable into the port with label 'PC'. Plug the other end into the Ethernet port of your PC. In doing this you will ensure that both the phone and your PC will be supplied with Internet without need of third device or second Internet line.
5. Power up the phone by plugging the power supply into the phone.
6. If you have headset plug it into the port with 'headset' image near it.
7. Attaching the Desk Stand
Note: Note that points 4 and 6 are optional.
Accounts Configuration
Prerequisitives
Before going further reading this tutorial make sure you have asterisk server and you are familiar with adding users to asterisk. If you are not aware of this issue you can read the tutorial explaining how to add new users to asterisk.
Linksys SPA942 is working with the SIP (Session Initiation Protocol). You will have to create a SIP user. Open /etc/asterisk/sip.conf and add the user at the bottom at the file. Since the hardphone support more than one lines we will create two accounts for the hardphone
[mytest1]
username=mytest1
secret=mytestsecret1
type=friend
host=dynamic
context=default
[mytest2]
username=mytest2
secret=mytestsecret2
type=friend
host=dynamic
context=other
username=mytest1
secret=mytestsecret1
type=friend
host=dynamic
context=default
[mytest2]
username=mytest2
secret=mytestsecret2
type=friend
host=dynamic
context=other
Then reload Asterisk's sip configuration by executing the command `asterisk -rx "sip reload"`.
At one point we will want to receive calls on our phone, that's why we have to create an extension for this. We will achieve this by adding extension `1001` in our default context in extensions.conf configuration file, located at /etc/asterisk. We will also create an extension `1002` which will dial the second account.
[default]
exten => 1001,1,Dial(SIP/mytest1,20)
exten => 1001,n,HangUp()
exten => 1002,1,Dial(SIP/mytest2,20)
exten => 1002,n,HangUp()
exten => 1001,1,Dial(SIP/mytest1,20)
exten => 1001,n,HangUp()
exten => 1002,1,Dial(SIP/mytest2,20)
exten => 1002,n,HangUp()
Of course we will have to reload our asterisk extension configuration in order the changes to take effect. We can easily achieve this by typing:
`asterisk -rx "extensions reload"` - for Asterisk 1.2.x
`asterisk -rx "dialplan reload"` - for Asterisk 1.4.x
`asterisk -rx "dialplan reload"` - for Asterisk 1.4.x
In order to use the voicemail features of the Linksys SPA942 we will create two mailbox accounts. In order to achieve this we will have to edit voicemail.conf. Here are the lines that we have to add:
[default]
11001 => 4242,Example Mailbox,myemail@server.com
11002 => 7474,Example Mailbox 2,myemail2@server.com
11001 => 4242,Example Mailbox,myemail@server.com
11002 => 7474,Example Mailbox 2,myemail2@server.com
You will have to make some modifications in the dialplan in order the voicemail to work properly. First well add Voicemail application in the part where we dial the two accounts.
exten => 1001,1,Dial(SIP/mytest1,20)
exten => 1001,2,VoiceMail(11001@default)
exten => 1001,n,HangUp()
exten => 1002,1,Dial(SIP/mytest2,20)
exten => 1002,2,VoiceMail(11002@default)
exten => 1002,n,HangUp()
exten => 1001,2,VoiceMail(11001@default)
exten => 1001,n,HangUp()
exten => 1002,1,Dial(SIP/mytest2,20)
exten => 1002,2,VoiceMail(11002@default)
exten => 1002,n,HangUp()
Then we will create two extensions which purpose will be to check the mailboxes accounts. Here are the lines that we are going to add to extensions.conf
exten => 111001,1,Answer()
exten => 111001,2,VoiceMailMain(11001@default)
exten => 111001,n,HangUp()
exten => 111002,1,Answer()
exten => 111002,2,VoiceMailMain(11002@default)
exten => 111002,n,HangUp()
asterisk rx reload
You can download the complete configuration files from our server:
Initialization
When you power up the phone it will start up. It will boot its operational system. When its ready you will be able to press the options button (the one which looks like page). In order to operate with the phone you will have to configure some options.
Network - network options are the the ninth option. Use the navigation buttons and press Select.
WAN Connection Type from this setting you can configure the type of the connection. The possible modes are: DHCP (dynamic address acquired from DHCP server), Static IP (using static settings) and PPPoE. In order to configure this, press the Edit button.
Current IP - this shows the current IP address of the phone.
Hostname - this shows the current hostname of the phone.
Domain - this shows the current domain of the phone.
Current Netmask - this shows the current network mask.
Current Gateway - this shows the current gateway.
Enable Web Server - by using the option you can configure if the web configuration is active or not
Non-DHCP IP Address - by using this option you can set static IP address. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP-Subnet Mask - by using this option you can set static network mask. Press the Edit button and enter a valid network mask. The asterisk (*) is used for dot.
Non-DHCP Default Route - by using this option you can set static gateway. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP DNS1 - by using this option you can set static primary DNS server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP DNS2 - by using this option you can set static secondary DNS server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP NTP Server1 - by using this option you can set static primary NTP server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP NTP Server2 - by using this option you can set static secondary NTP server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Multicast Address - by using this option you can set static multicast Address. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Enable VLAN - enable this parameter if the switch to which the Linksys SPA942 is connected to uses VLAN tagging.
VLAN - enter the VLAN tag which the Linksys SPA942 is assigned to.
CDP - enable or disable the Cisco Discovery Protocol.
Current IP - this shows the current IP address of the phone.
Hostname - this shows the current hostname of the phone.
Domain - this shows the current domain of the phone.
Current Netmask - this shows the current network mask.
Current Gateway - this shows the current gateway.
Enable Web Server - by using the option you can configure if the web configuration is active or not
Non-DHCP IP Address - by using this option you can set static IP address. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP-Subnet Mask - by using this option you can set static network mask. Press the Edit button and enter a valid network mask. The asterisk (*) is used for dot.
Non-DHCP Default Route - by using this option you can set static gateway. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP DNS1 - by using this option you can set static primary DNS server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP DNS2 - by using this option you can set static secondary DNS server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP NTP Server1 - by using this option you can set static primary NTP server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Non-DHCP NTP Server2 - by using this option you can set static secondary NTP server. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Multicast Address - by using this option you can set static multicast Address. Press the Edit button and enter a valid IP address. The asterisk (*) is used for dot.
Enable VLAN - enable this parameter if the switch to which the Linksys SPA942 is connected to uses VLAN tagging.
VLAN - enter the VLAN tag which the Linksys SPA942 is assigned to.
CDP - enable or disable the Cisco Discovery Protocol.
SIP Options - in order to configure the account on the phone you will have to use the web-interface. In order to have access to it you will need to configure you network settings and enable the web interface from the phone menus (it is enabled by default). Type the IP address in your browser and enter into administration mode by clicking `Admin Mode`. You will be prompted for password if you have set such. Then go to Ext 1 tab as is shown below.
This is the location where you can configure your account.
Enable the line by selecting `Line Enable` option to yes
Check if there is NAT between your hard phone and the Asterisk server and if so set appropriate NAT options
If you are debugging set Debug mode.
Enter the IP address or the hostname of your Asterisk server on the `Proxy` setting
Set `User ID` and `Auth ID` to the account name of the SIP account that you created
Set the password for the account
Select appropriate audio codec
To apply changes press `Submit All Changes` button and reboot the phone.
Check if there is NAT between your hard phone and the Asterisk server and if so set appropriate NAT options
If you are debugging set Debug mode.
Enter the IP address or the hostname of your Asterisk server on the `Proxy` setting
Set `User ID` and `Auth ID` to the account name of the SIP account that you created
Set the password for the account
Select appropriate audio codec
To apply changes press `Submit All Changes` button and reboot the phone.
Repeat this operation with the second SIP account that we created (mytest2) but on Ext 2
Now you are ready to use your Linksys SPA942. In the next pages you will find some advances options and configurations of the hardphone.
Using the Web interface
Most of the options are accessible through the web configuration interface. In order to access the web interface you will need a correctly configured network. An example is shown on the previous page.
The web interface is divided into seven tabs. Every tab gives different information. Here are the eight tabs:
The first tab named `Info` will show you different information for your hardphone.
System Information - this part will give you detailed info for your network configuration such as DHCP info, Current IP, Hostname, Domain, Current Netmask, Current Gateway, Primary and Secondary DNS.
Product Information - this part will give you detailed info for your product such as - Product Name, Serial Number, Software & Hardware Version, MAC Address, Client Certificate and Licenses.
Phone Status - this part will give you info on some of the phone statistics such as - Current Time, Uptime, Broadcast, RTP & SIP Packets Statistics
Ext 1 Status - this part will give you information on registration status of the phone - Registration State, Last Registration Time, Next Registration Timing, Message Waiting, Mapped SIP Port
Ext 2 Status - this part will give you information on registration status of the phone - Registration State, Last Registration Time, Next Registration Timing, Message Waiting, Mapped SIP Port
Line 1 Call 1 Status - this part will give you detailed information on current call that is been made through Line 1. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Line 1 Call 2 Status - this part will give you detailed information on current call that is been made through Line 1. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Line 2 Call 1 Status - this part will give you detailed information on current call that is been made through Line 2. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Line 2 Call 2 Status - this part will give you detailed information on current call that is been made through Line 2. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Product Information - this part will give you detailed info for your product such as - Product Name, Serial Number, Software & Hardware Version, MAC Address, Client Certificate and Licenses.
Phone Status - this part will give you info on some of the phone statistics such as - Current Time, Uptime, Broadcast, RTP & SIP Packets Statistics
Ext 1 Status - this part will give you information on registration status of the phone - Registration State, Last Registration Time, Next Registration Timing, Message Waiting, Mapped SIP Port
Ext 2 Status - this part will give you information on registration status of the phone - Registration State, Last Registration Time, Next Registration Timing, Message Waiting, Mapped SIP Port
Line 1 Call 1 Status - this part will give you detailed information on current call that is been made through Line 1. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Line 1 Call 2 Status - this part will give you detailed information on current call that is been made through Line 1. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Line 2 Call 1 Status - this part will give you detailed information on current call that is been made through Line 2. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
Line 2 Call 2 Status - this part will give you detailed information on current call that is been made through Line 2. Info includes: Call State, Tone, Encoder & Decoder, Type, Hold, Callback, Peer's Name & Phone, Duration, Packets Sent & Received, Bytes Sent & Received, Decode Latency, Jitter, Round Trip Delay, Packets Lost, Packet Error, Mapped RTP Port
The second tab named `System` will allow you to reconfigure hardphone's network configuration.
System Configuration - from this part you will be able to set the user's password and Web Server status (Enabled/Disabled)
Internet Connection Type - from this part you will be able to set basic network options such as - DHCP, Static IP, Network Mask and Gateway
Optional Network Configuration - from this part you will be able to set additional network options such as - HostName, Domain, Primary & Secondary DNS, DNS Query Mode, Syslog & Debug Server, Debug Level
Internet Connection Type - from this part you will be able to set basic network options such as - DHCP, Static IP, Network Mask and Gateway
Optional Network Configuration - from this part you will be able to set additional network options such as - HostName, Domain, Primary & Secondary DNS, DNS Query Mode, Syslog & Debug Server, Debug Level
The third tab named `SIP` will allow you the configure some advanced SIP options
Response Status Code Handling - this page will allow you to set the code value for SIT1 RSC, SIT2 RSC, SIT3 RSC and SIT4 RSC
RTP Parameters - this page will allow you to set the port range for RTP.
SDP Payload Types - this page will allow you to set payload values for: AVT Dynamic Payload, G726r16 Dynamic Payload, G726r24 Dynamic Payload, G726r40 Dynamic Payload, G729b Dynamic Payload
RTP Parameters - this page will allow you to set the port range for RTP.
SDP Payload Types - this page will allow you to set payload values for: AVT Dynamic Payload, G726r16 Dynamic Payload, G726r24 Dynamic Payload, G726r40 Dynamic Payload, G729b Dynamic Payload
The fourth tab named `Regional` will allow you to configure some Regional settings as well as the service activation and deactivation codes
Control Timer Values - this part will allow you to set the timing in seconds for the following services: Call Back Expires and Call Back Retry Interval
Vertical Service Activation Codes - this page will allow you to set the codes for some of the services - Call Return, Blind Transfer, Call Back Activation & Deactivation, Call Forward All Activation & Deactivation, Call Forward Busy Activation & Deactivation, Call Forward No Answer Activation & Deactivation, Block CallerID Activation & Deactivation, Block CallerID Per Call Activation & Deactivation, Block ANC Activation & Deactivation and DND Activation & Deactivation
Miscellaneous - from this part you will be able to set some regional settings such as - Local Date, Local Time, Time Zone and Time Offset
Vertical Service Activation Codes - this page will allow you to set the codes for some of the services - Call Return, Blind Transfer, Call Back Activation & Deactivation, Call Forward All Activation & Deactivation, Call Forward Busy Activation & Deactivation, Call Forward No Answer Activation & Deactivation, Block CallerID Activation & Deactivation, Block CallerID Per Call Activation & Deactivation, Block ANC Activation & Deactivation and DND Activation & Deactivation
Miscellaneous - from this part you will be able to set some regional settings such as - Local Date, Local Time, Time Zone and Time Offset
From the fifth tab named `Phone` you will be able to set some settings concerning the user such as sounds, naming, logos, etc...
General - from this part you will be able to set some basic customization such as: Station Name, Voice Mail Number and Text Logo
Ring Tone - from this part you will be able to set ten custom ringing types
Audio Input Gain - from this part you will be able to set some options that will alter the gain of sound for the different microphones. This includes - Handset, Headset and Speakerphone. You will be able to set plus/minus six dB on each of them
Ring Tone - from this part you will be able to set ten custom ringing types
Audio Input Gain - from this part you will be able to set some options that will alter the gain of sound for the different microphones. This includes - Handset, Headset and Speakerphone. You will be able to set plus/minus six dB on each of them
From the sixth and seventh tabs named `Ext 1` and `Ext 2`you will be able to set the SIP settings for Ext 1 and Ext 2
General - from this part you will be able to set if the Line is Enabled or Not
NAT Settings - from this part you will be able to set if the NAT Mapping and NAT Keep Alive are Enabled or Not
SIP Settings - from this part you will be able to set custom SIP Port and SIP Debug Options
Call Feature Settings - from this part you will be able to set Message Waiting, Default Ringing and the Mailbox ID
Proxy and Registration - from this part you will be able to set SIP Proxy Server (the IP address or hostname of the Asterisk server), Register Expiration Timing as well as Making and Receiving Calls Without Registration
Subscriber Information - from this part you will be able to set you SIP account settings such as: Display Name, User ID, Password and Auth ID
Audio Configuration - from this part you will be able to set some audio settings such as Preferred Codec, Usage of Different Codecs, Silence Suppression and DTMF Tx Method
NAT Settings - from this part you will be able to set if the NAT Mapping and NAT Keep Alive are Enabled or Not
SIP Settings - from this part you will be able to set custom SIP Port and SIP Debug Options
Call Feature Settings - from this part you will be able to set Message Waiting, Default Ringing and the Mailbox ID
Proxy and Registration - from this part you will be able to set SIP Proxy Server (the IP address or hostname of the Asterisk server), Register Expiration Timing as well as Making and Receiving Calls Without Registration
Subscriber Information - from this part you will be able to set you SIP account settings such as: Display Name, User ID, Password and Auth ID
Audio Configuration - from this part you will be able to set some audio settings such as Preferred Codec, Usage of Different Codecs, Silence Suppression and DTMF Tx Method
From the last tab named `User` you will be able to set additional user options such as Call Forwarding, Speed Dialing and Audio settings.
Call Forward - from this part you can set user axiomatizations which concerns call forwarding and dialing - Call Forward All Destination, Call Forward Busy Destination, Call Forward No Answer Destination, Call Forward No Answer Delay, Speed Dial numbers (2 to 9)
Supplementary Services - from this part you can set miscellaneous user options such as: Block CallerID, Block ANC, DND, Secure Call, Dial Assistance, Auto Answer Page, Preferred Audio Device, Time Format, Date Format
Audio Volume - from this part you can audio volume for: Ringer, Speaker, Handset and Headset
Supplementary Services - from this part you can set miscellaneous user options such as: Block CallerID, Block ANC, DND, Secure Call, Dial Assistance, Auto Answer Page, Preferred Audio Device, Time Format, Date Format
Audio Volume - from this part you can audio volume for: Ringer, Speaker, Handset and Headset
Links
Here we have gathered some useful links and resources:
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Latest Headlines:
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T.38 faxing with Zoiper 2.15 is now easier than ever
section: voip software
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Asterisk 1.4.21 Released
section: Asterisk
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Asterisk 1.4.20 Released
section: Asterisk
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Asterisk 1.4.20-rc2 Released
section: Asterisk
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Asterisk 1.4.20-rc1 Now Available
section: Asterisk
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Latest Tutorials:
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Sending Fax from Zoiper to Zoiper using T.38
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VMAuthenticate (dialplan application)
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added 06/Nov/2007 17:57
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Siemens C455 IP hardphone
added 05/Nov/2007 10:24
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Zoiper
added 22/Oct/2007 17:53
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