Configure Cisco SPA512G

In the article “Configure Cisco SPA512G” 📱 you will find an administrator’s guide that will help you configure SPA512G with the most complex functions of your Cisco terminal. 📖 Learn everything you need to know for optimal setup!

This SPA512G configuration guide will help you to optimally configure and use the terminal. If you are interested in knowing how to use the most basic functions of the terminal, we invite you to consult the Manual Cisco SPA512G user level that will teach you among many other things to:

• • Transfer calls
  • Make Conferences
  • Divert calls
  • Mute phone

And a whole lot more!

How to configure Cisco SPA512G and update firmware?

Phones must be upgraded to the latest firmware before using any management features. There are several ways to update your firmware, depending on the phone model and the call control system the phone is used with.

How to know the Firmware version we have on Cisco SPA512G?

To determine the current firmware version:

Step 1 Press the Configure 

Step 2 Scroll to Product Information and press Select. The current firmware is displayed under Software version.

How to know the IP address on Cisco SPA512G?

Before upgrading the device, you must know the IP address of the phone you are upgrading. Often, a DHCP server assigns an IP address, so the phone must boot up and connect to the subnet. .

To display its IP address:

Step 1 Press the Settings 

Step 2 Scroll to Network and press Select. The IP address is displayed in Current IP.

Where to download firmware for Cisco SPA512G?

To download the firmware from Cisco.com to your PC:

Step 1 Direct your browser to the URL http://www.cisco.com/cisco/software/navigator.html?a=a&i=rpm.

Step 2 Follow the instructions on the page to locate your product and download the firmware file. 

Step 3If the firmware file you download is in zip format, double-click the file and extract its contents to a single folder or to the desktop. 

How to install the firmware on Cisco SPA512G?

Your computer must be on the same subnet as the phone you are upgrading.

Step 1 With your PC connected to the same subnet as the phone, run the executable file for the firmware update.

Step 2 Click Continue after reading the message about the update and your service provider’s information.

Step 3 Enter the IP address of the phone.

Step 4 Follow the instructions on the screen.

How to upgrade the firmware on Cisco SPA512G?

• Firmware upgrade executable file: download the firmware upgrade utility from the related products page at Cisco.com to your PC. Run the update by double-clicking on the executable file. Your computer must be on the same subnet as the Cisco SPA IP Phones.
• Auto-provisioning: a phone downloads a configuration file that includes firmware upgrade information when it is powered on or configured to do so. The configuration file (also known as a profile) includes parameters that indicate how and when the phone’s firmware should be updated. 

How to save firmware upgrade changes on Cisco SPA512G?

Click Send all changes when you have finished modifying the fields in the phone’s web user interface to update the configuration profile. The phone restarts and the changes are applied.

Click Undo all changes if you want to clear all changes made in this session and revert to the parameter values set before the session started or since the last time you clicked Send all changes.

How to access the Administrator interface on Cisco SPA512G?

Access the administrator options in any of the following ways:

• Log in to the configuration utility, then click Admin Login.

• Enter the IP address of the phone in a web browser and include the admin/extension. For example: http://192.168.1.220/admin/

How to configure lines on Cisco SPA512G?

Cisco SPA IP phones offer different numbers of lines depending on the phone model. Each line corresponds to a phone number (or extension) used for calls.

Each line can support two calls. For example, a four-line phone can handle eight calls. One call can be active (in conversation) and seven can be on hold.

This chapter contains the following sections:

• Setting a line key
• Assigning busy indicator light, call pickup or speed dial functions to unused lines
• Configuration of audio indication for call pickup event
• Basic configuration or Advanced function of call parking to a line key

How to configure a line key on the Cisco SPA512G?

Each line key can be assigned multiple extensions, a short name, and a shared call appearance. The number of line keys depends on the IP phone model. Generally, you should reserve line key 1on the IP phone as the designated user’s primary, private extension.

To set up a phone line:

Step 1 Click administrator session > advanced > Voice > Phone.

Step 2 Under each line key on the phone, configure the following:

• Extension: assign an extension number to the line key.Default value is 1.
• Short name: enter a short name or number to display on the IP phone screen.
• Share call appearance: select shared to share incoming call appearances with other phones. See Shared line appearance settings. If you select private, the calling line is not shared with any other phone. Default to private.
• Extended feature: see Assigning busy indicator light, call capture or speed dial field features to unused lines.

Step 3 Click Submit all changes.

How to configure the appearance of a shared line key on Cisco SPA512G?

You create a shared line appearance by assigning the same directory number to different devices. A Cisco system considers a directory number to be a shared line if it appears on more than one device.

In a shared call appearance, for example, you can configure a shared line so that a directory number appears on line 1 of an administrator’s phone and line 2 of an attendant’s phone.

Another example of a shared line involves a single incoming 800 number that is configured to appear as line 2 on each sales rep’s phone in an office. 

Most devices with a shared line appearance can make or receive new calls or resume waiting calls at the same time.

Incoming calls are displayed on all devices sharing a line and anyone can answer the call. Only one call remains active at a time on a device. 

Call information (such as caller or called party) is displayed on all devices sharing a line.

If one of the devices activates the Privacy feature, the other devices sharing the line will not see outgoing calls made from the device that activated privacy. All devices will still see incoming calls to the shared line. 

Devices with shared call appearances can initiate separate transfers or conference transactions. 

When a call is placed to the extension number for the shared call, all shared Cisco SPA IP phones ring. Any IP phone can answer the call. S

if the active phone places the shared call on hold, the call can be resumed from any of the shared Cisco SPA IP Phones by pressing the corresponding line key or the Selectwhen Resume icon is displayed. 

The Cisco SPA500 Series IP Phones support private standbyfor MetaSwitch and BroadSoft. Users who have a shared line can press PrivHoldand only the user who placed the call on hold can resume the call..

Each IP phone can be configured independently. Although the account information is generally the same for all Cisco SPA IP phones, settings such as dial plan or preferred codec can vary between phones and still support the shared line appearance.

To configure the line:

Step 1: Click Admin Login > advanced > Voice.

Step 2: Click on the Ext_nof the extension being shared (do not use Ext 1).

Step 3: In General in the Enable line list, choose yes.

Step 4: In Share Line Appearance in the Share Ext list, choose shared. If you set this extension as private (not shared), the extension does not share calls, regardless of the Share Call Appearance on the Phone tab.

If you configure this extension as shared, calls follow the Call Appearance Share on the Phone tab. On Cisco SPA500 series phones that have line buttons, a hollow phone icon is displayed next to the shared line button.

Step 5: In the Shared User ID field, enter the user ID (name) of the phone with the extension that is being shared.

Step 6: In the Subscription Expires field, enter the number of seconds before the SIP subscription expires. Until the subscription expires, the phone receives NOTIFICATION messages from the SIP server about the status of the shared phone extension. The default value is 60 seconds. 

Step 7: In the Restrict MWI (message waiting indicator), choose yesto set the message waiting indicator to light only for messages on private (SIP) lines. Choose no to set the message waiting indicator to light for all messages..

Step 8: Under Proxy and Logging, in the Proxy , enter the IP address of the proxy server (for example, the Cisco SPA9000 IP address).

Step 9: Under Subscriber Information, enter a Display Nameand User ID (extension number) for the shared extension. These are displayed on the IP phone screen..

Step 10: (Optional) On the Phone Miscellaneous Line Key Settings tab, configure SCA Barge-In Enable. Choose yes to allow users sharing call appearances to take over the call on a shared line. Choose no to prevent users from taking over the call on a shared line. 

For example, Bob and Chris share extension 401. Adam calls extension 401. Bob answers the call. Adam and Bob are connected.

If Chris has the Enable SCA Interrupt field on his phone set to yes, he can press the line button for extension 401. Chris and Adam are connected on a call and Bob is disconnected from the call.

Step 11: Click Submit all changes.

How to configure call appearance per line with Cisco SPA512G?

On the Phone Line Call Appearance tab (under Miscellaneous Line Key Configuration) allows you to choose the number of calls per line button. The default value is 2. 

When you increase the number of calls per line to a value greater than 2, you must configure the following:.

• Line ID assignment (in Miscellaneous Line Key Settings) under Horizontal First.

• Line navigation (in Miscellaneous line key setting) to By call.

• Enable (in Programmable keys) to Yes.

When the maximum number of calls per phone is reached, the phone does not allow you to place a new call and rejects incoming calls. On Cisco SPA512G the maximum number of calls allowed is 10.

How to expand the call appearance per line in Cisco SPA512G?

To expand the per-line call appearances:.

Step 1 Click admin session > advanced > Voice > Telephone.

Step 2 In the Miscellaneous Line Key Settings in the Call Appearance Per Line field, choose how many calls per line to allow from the drop-down menu.

How to configure the call appearance mapping style in Cisco SPA512G?

On the Phone tab, Miscellaneous Line Key Settings, allows you to configure the line assignment. Each LED (line/extension) can contain 2 calls (default)..

You can assign an extension to both line LEDs. The first call always causes the assigned LED to flash. Choose one of the following: .

• First vertical next phone LED flashes with the second incoming call.

• First horizontal: the same LED flashes with the second incoming call.

How to configure unused line keys to access services on Cisco SPA512G?

On Cisco SPA500 Series IP Phones, unused or idle phone lines can also be configured to access services, such as: .

• XML services

To configure the line keys to access the services:

Step 1 Click Admin Login > advanced > Voice > Phone.

Step 2 On the line key to configure (line 4 in this example):.

• In the Extension drop-down list
• Enter the following string in the Extended Function field: fnc=type where:

• • fnc:

• • function-type: .
    • xml: pressing the line button accesses the XML services. The XML service configured in the Phone tab below opens the page identified in the XML Service field (see Configuring XML Services). You can specify a different XML service to connect to using the syntax fnc=xml;URL=http://xxx.xx.xxx/entry.html, where xxx.xx.xxx is the URL of the XML service.
    • mp3: clicking on the line button starts the mp3 player.

• • • weather: pressing the line button accesses the weather information.

• • • news: pressing the line button accesses the news.

For example, to configure line 4 for mp3 player:

fnc=mp3.

Step 3 Click Send all changes. After restarting the phone, the configured lines light up orange and display the following icons next to the extension label:.

• xml: XML icon
• news: RSS icon
• weather: thermometer icon

How to configure BroadSoft advanced park and retrieve call support on Cisco SPA512G?

You can configure the SPA phone to park a call from another phone and cancel the call from any other phone.

For detailed instructions on how to configure phones with BroadSoft’s call park and unpark feature, see Configuring Call Parking with Broadsoft, available from the Cisco support community at https://supportforums.cisco.com/docs/DOC-15426.

BroadSoft Call Park and Recall enhances the caller experience by:

• Enabling front desk staff to distinguish between a reversed call and a new call.

• Provide additional call routing and handling options that improve front office efficiency when the receptionist is unavailable or busy.

• A visual indication is provided to users when a call is parked at their extension.

For detailed signaling requirements, refer to the Broadsoft/Cisco Partner Configuration Guide.

How to perform assignment, capture, and monitoring of a parked call on Cisco SPA512G?

Users can assign a dedicated call park button to park a call, capture a parked call, and monitor the park status. The park number has to be the SIP line configured on the phone.

To assign and answer a parked call:

Step 1 Click admin session > advanced > Voice > Phone

Step 2 Choose Line key to configure (line 5 in this example).

• In the Extension drop-down list
• In the Share call appearance drop-down list Private.
• Enter the following string in the Extended function : fnc=prk;sub=1388@$PROXY;vid=1 where:

• • fnc: function
  • prk: call parking
  • sub: parking subscription account
  • orbit: parking lot number or location where the call is parked.

Note On a shared line, the subscriber’s user ID may be different from the shared account number. The user can use the “orbit” keyword to define the parking number (usually it is the shared number).

For example, if the shared user ID is 1391-user2 and the shared number is 1391, in the Extended Function :

fnc=prk;sub=1391-user2@$PROXY;orbit=1391;vid=2

Step 3 Click on Console.

Note Set the appropriate access codes for Broadsoft’s call parking feature (Operator Console Call Parking Code) and call capture (Operator Console Call Deskew Code).

fnc=prk;sub=1391-user2@$PROXY;orbit=1391;vid=2

Step 4 Click submit all changes.

How to configure unused keys on Cisco SPA512G?

You can configure unused or inactive lines on Cisco SPA500 Series IP Phones to interact with another line in the system.

For example, if you have two inactive lines on an attendant’s phone, you can configure those lines to display the status of a supervisor’s phone (Busy Light Field [BLF]).

You can also configure the idle lines so that they can be used to speed dial the supervisor’s phone, or take calls that are ringing on the supervisor’s phone.

A monitored extension can be either private or shared.

How to configure call capture and busy light field on Cisco SPA512G?

You must enable BLF to configure call capture.

In this example, Attendant Bob (extension 200) has an idle line (line 4) on his Cisco SPA508G. He would like to be able to see if his supervisor Stephanie (extension 300) is on the phone and take the calls that are ringing on her extension.

To set up this feature for Bob:

Step 1 Click admin session > advanced > Voice > Telephone.

Assigning busy light, call pickup or speed dial functions to unused lines.

Step 2 On the line key to configure (line 4 in this example):.

• the Extension drop-down list
• Enter the following string in the Extended Function field: fnc=blf+cp;sub=Stephanie@$PROXY;ext=300@$PROXY Using the following syntax: fnc=type;sub=stationname@$PROXY;ext=extension#@ $PROXY where:
  • fnc: function
  • blf: busy light field
  • cp: call capture
  • sub: station name
  • ext or usr: extension or user (the usr and ext are interchangeable)

Step 3 Click Send all changes. After the phone reboots, the phone in this example displays the following LED colors for the monitored lines:.

• Green: Available
• Red: Occupied
• Red Rapidly flashing: Ringing

If the phone’s LEDs display orange or flash orange slowly, there is a problem: Orange indicates that the phone failed to subscribe (received a 4xx response) and flashing orange slowly indicates an undefined problem (there may be no response to the subscription request or BLF).

When the phone is set up correctly, Bob can monitor Stephanie’s line. When a call rings on Stephanie’s line, he can press the line 4 button to answer it.

If you set the BLF call capture feature on a programmable line key, the user can select any register line key explicitly to capture the call, regardless of the “vine” setting.

How to configure speed dial on Cisco SPA512G?

In this example, the attendant, Bob (extension 200), has another idle line (line 5) on his Cisco SPA508G. He wants to quickly dial his supervisor Mark (extension 400) from that line.

To configure this feature for Bob:

Step 1 Click admin session > advanced > Voice > Telephone.

Step 2 On the line key to configure (line 5 in this example):.

• the Extensiondrop-down list
• In the Share call appearance drop-down list, choose private.
• Enter the following string in the Extended Function field: fnc=sd;ext=400@$PROXY Using the following syntax:

fnc=type;ext=extension#@@$PROXY

where:

• • fnc: function
  • sd: speed-dial
  • ext or usr: extension or user (the usr and ext are interchangeable)

Step 3 Click Send all changes. 

When the phone is set up correctly, Bob can press the line 5 button to dial Mark’s line.

How to configure DTMF hold and pause settings on Cisco SPA512G?

Speed dial, directory, extended feature, and other strings configured on the phone can include standby (X) and pause (,) that allow manual and automatic transmission of DTMF (dual-tone multi-frequency) signals. 

Syntax: 

{Dial_String}[ ][,|X][DTMF_String][,|X][DTMF_string] Where:

Dial_Stringnumber the user is trying to reach. For example, 8537777 or 14088537777.

(space): a dial termination that defines or delimits the end of the dial string. The space is mandatory. If the phone encounters an X or a comma (,) before the space, the characters are treated as part of the dial string.

, (comma): a 2-second pause that is inserted for each comma in the string. 

X (wait): waits for user input and acknowledgement.

• The message “Waiting for further digit input. When finished, press OK to continue”message appears. 

• The user manually enters DTMF signals using the dial pad. 

• The user selects OK, acknowledging that the manual entry transmission is complete. 

• The phone sends any DTMF signal defined by DTMF_string. 

• The phone executes the next parameter. If there are no more parameters in the dial string to execute, the phone exits to the main screen. 

The hold does not time out; the hold message window is not dismissed until the user confirms the hold message or the call is terminated by the user or by the remote device.

DTMF_string:DTMF signals that a user sends to a remote device after the call is connected. The phone cannot send signals that are not valid DTMF signals. 

Example.

18887225555,,5552X2222

A speed dial entry causes the phone to dial 18887225555. The space indicates the end of the dial string. The phone waits 4 seconds (2 commas) and then sends DTMF signals 5552. 

A message is displayed prompting the user to manually dial DTMF signals. When the user has finished dialing DTMF signals, the user selects OK to confirm that the manual DTMF signal transmission is complete. The phone sends DTMF signals 2222. 

Guidelines for use.

A user can transmit DTMF signals at any time, as long as the call is connected. 

The length of the string, including X’s or commas (,), that can be supported is limited to the length of a speed dial entry, a dial screen entry, a directory entry, etc. 

If the screen displayed is not a call appearance screen and a hold is initiated, the phone displays the home screen and prompts the user to enter more DTMF signals. If this occurs while the user is editing an entry, edits may be lost..

If only the first part of a dial string matches a dial plan, when the call is dialed, the part of the dial string that does not match the dial string is ignored. For example:.

8537777776666 ,,1,23

If 8537777777matches a dial plan, the 666666characters are ignored. The phone waits 4 seconds before sending DTMF 1. Then it waits 2 seconds and sends DTMF 23..

When registering the call, the phone registers only the dial string; DTMF strings are not registered. 

Valid DTMF signals are 0-9, * or #. All other characters are ignored. 

Limitations

When the call is connected and transferred immediately, the phone may not be able to process DTMF signals, depending on the duration of the call connection before transferring the call. As soon as the call is connected, it is transferred. 

This feature is not supported when the phone is in SPCP mode. 

How to configure audio indication for call capture event on Cisco SPA512G?

You can configure the phone to play the call capture tone when there are incoming calls on any of the lines that the user is monitoring with the call capture feature.

To set the audio prompt:

Step 1 Click Admin Login > advanced > Voice > Att(endant) Console. .

Step 2 In the General section Yes.

To configure this parameter via the configuration file, set the following line for the profile:.

<Call_Pickup_Audio_Notification ua=”na”>Yes.

</Call_Pickup_Audio_Notification>

Step 3 Click regional 

Step 4 In the Ongoing Ringtones section Tone parameter.

The default value is 440@-10;30(.3/9.7/1), same as the ringing tone on hold.

To set this parameter via the configuration file, configure the following profile line: <Call_Pickup_Tone ua=”na”>440@-10;30(.3/9.7/1)</Call_Pickup_Tone>.

Step 5 Click Send all changes. 

How to configure the basic or advanced call park feature for a line key on Cisco SPA512G?

Unused line keys can be enabled to allow call parking (for MetaSwitch software switch) on Cisco SPA500 series phones. Users can press this line button to park a call or retrieve a parked call.

Step 1 Click admin session > advanced > Voice > Attention (extreme) Console. 

Step 2 In the Generalsection under Server Type, choose RFC3265_4236.

Step 3 Click on the Phone tab.

Step 4 Choose the line key to configure (line 5 in this example):.

• In the Extensiondrop-down list
• In the Share call appearancedrop-down list
• Enter the following string in the Extended function field: fnc=prk;[email protected] where:

• • fnc: function
  • prk: call park
  • sub: call parking orbit or location where the call is parked. The valid range of values is from 01 to 10. In this example, 5 is used.
  • domain.com: domain of the phone, usually the same as the proxy on the Ext 1 tab. You can also use fnc=prk;sub=05@$PROXY to set this value.

Step 5 Click Submit all changes. 

How to configure SIP, SPCP and NAT on Cisco SPA512G?

Cisco SPA IP Phones use the following protocols:

• Session Initiation Protocol (SIP): Cisco SPA500 Series
• Cisco Smart Phone Control Protocol (SPCP): Cisco SPA500 Series

This chapter describes how to configure the telephone protocols: .

• SIP and Cisco IP phones,
• SIP configuration
• Communications protocol configuration of the IP phone
• Protocol configuration on a Cisco SPA500 series IP Phone
• Administration of NAT Traversal with Cisco IP Phones

How to manage SIP specifications on Cisco SPA512G?

Cisco IP phones use the Session Initiation Protocol (SIP), which allows interoperation with all IT service providers that support SIP.

SIP handles signaling and session management within a packet telephony network. Signaling allows call information to be transmitted across network boundaries. Session management controls the attributes of an end-to-end call.

The diagram shows a SIP request for connection to another subscriber on the network. 

In typical commercial IP telephony deployments, all calls go through a SIP proxy server. The requesting phone is referred to as the SIP user agent server (UAS), while the receiving phone is referred to as the user agent client (UAC). 

SIP message routing is dynamic. If a SIP proxy receives a request from a UAS for a connection but cannot locate the UAC, the proxy forwards the message to another SIP proxy on the network.

When the UAC is located, the response is routed back to the UAS and a direct peer-to-peer session is established between the two UAs. Voice traffic is transmitted between UAs over dynamically assigned ports using Real-Time Protocol (RTP). 

RTP transmits real-time data such as audio and video; it does not guarantee real-time data delivery. 

RTP provides mechanisms for sending and receiving applications to support data transmission. Typically, RTP runs over UDP. See NAT mapping with STUN.

How to manage SIP over TCP on Cisco SPA512G?

To ensure stateful communications, Cisco IP phones can use TCP as the transport protocol for SIP.

This protocol provides guaranteed deliverywhich ensures that lost packets are retransmitted. TCP also ensures that SIP packets are received in the same order in which they were sent. 

TCP overcomes the problem of UDP ports being blocked by corporate firewalls. With TCP, there is no need to open new ports or discard packets, because TCP is already in use for basic activities, such as Internet browsing or e-commerce. 

How to manage SIP proxy redundancy on Cisco SPA512G?

An average SIP proxy server can handle tens of thousands of subscribers. A backup server allows you to temporarily take an active server offline for maintenance. Cisco phones support the use of backup SIP proxy servers to minimize or eliminate service interruption.

A static list of proxy servers is not always adequate. If your user agents are served by different domains, for example, you would not want to configure a static list of proxy servers for each domain on each Cisco IP phone.

A simple way to support proxy redundancy is to configure a SIP proxy server in the Cisco IP Phone configuration profile. DNS SRV records instruct phones to communicate with a SIP proxy server in a domain named in SIP messages.

The phone queries the DNS server. If configured, the DNS server returns an SRV record containing a list of SIP proxy servers for the domain, with their host names, priority, listening ports, etc. The Cisco IP Phone attempts to communicate with the hosts in the order of their priority..

If the Cisco IP Phone currently uses a lower priority proxy server, the phone periodically polls the higher priority proxy and switches to the higher priority proxy when it is available.

How to configure SRST support on Cisco SPA512G?

The proxyand outbound proxyin the Ext can be configured with an extension that includes a statically configured DNS SRV record or a DNS A record. This allows failover and backup functionality with a secondary proxy server. The format for the parameter value is:.

FQDN format: hostname [: port] [: SRV = host-list OR: A = IP-list] host-list: srv [| srv [| srv …]] srv: hostname [: port][:p=priority][:w=weight][:A=ip-list] ip-list: ip-addr[,ip-addr[,ip-addr[,ip-addr…]]

The default priority is 0 and the default weight is 1. The default port is 0 and the application substitutes the appropriate port value (e.g., port 5060 for SIP).

How to manage dual registration on Cisco SPA512G?

The phone always registers to both the primary (or primary outbound) proxy and the alternate (or alternate outbound) proxy. After registration, the phone sends invite and non-invite SIP messages through the primary proxy first.

If there is no response for the new INVITE from the main proxy, after the timeout, the phone should try the alternate proxy..

Dual registration per line is supported. Three new parameters were added that can be configured via the web GUI and remote provisioning:

• Alternate proxy: default is empty
• Alternate output proxy: default value is empty
• Dual registration: default is NO (disabled)

When configuring the settings, reboot the phone for the feature to take effect.

Note The administrator must specify a value for the primary proxy (or primary outbound proxy) and alternate proxy (or alternate outbound proxy) for the feature to work correctly.

What are the limitations for dual registration and DNS SRV redundancy on Cisco SPA512G?

As of this release (7.5.6), the limitations for dual registration and DNS SRV redundancy are as follows:

• When dual registration is enabled, DNS SRV DNS proxy retrieval/redundancy must be disabled.
• Do not use dual registration in conjunction with other backup/recovery mechanisms. For example: Broadsoft mechanism.
• When dual logging is enabled, the Automatic logging when failover parameter must be disabled.
• There is no recovery mechanism for the feature request. However, the administrator can adjust the re-registration time for a quick update of the registration status for the primary and alternate proxy.

How to manage alternate proxy and dual registration on Cisco SPA512G?

When the Dual Registration parameter is set to no alternate proxy, it is ignored.

How to perform registration after failover/recovery?

• Recovery error: the SPA phone performs failover to the secondary proxy when the SIP request gets no response from the primary proxy.

• Recovery: the phone attempts to re-register with the primary proxy while it is registered or actively connected to the secondary proxy.

Note In version 7.5.5 and earlier versions, to control re-registration in case of failover or recovery, “BT” was configured in the parameter <Test RSC backup>. From version 7.5.6 onwards, this is replaced by another new parameter <Auto-register when failover>.

Previous behavior (7.5.5 and earlier):

In the failover/recovery scenario, the phone remains registered on the primary proxy, but makes and receives calls on the alternate proxy.

The SPA IP phone is re-registered in case of failover or recovery with “BT” configured in the <Test RSC backup> field in the “Answer Status Code Handling” section of the SIP tab.

• Note: The <Try Backup RSC> parameter is used to invoke failover upon receipt of the specified response codes.

New behavior (7.5.6):.

When the new parameter <Automatically log when failover occurs> is set to yes, the phone cancels the logging and logs as the previous behavior (when a special value “BT” is set in the parameter <Test RSC backup>.

The solution is to remove the definition “BT” and add the new parameter, <Auto Register When Failover> to control the phone to automatically register to the failover proxy when the failover action is executed The INVITE message will be forwarded to the new proxy.

How to manage fallback behavior on Cisco SPA512G?

When the Auto Register When Failover parameter is set to no, fallback occurs immediately and automatically. If Proxy Fallback Intvl exceeds, all new SIP messages go to the main proxy.

When the parameter, Automatic Registration When Failover is set to Yes, the fallback occurs only when the current registration expires, which means that only the REGISTER message can trigger the fallback.

. the backup is activated 3600 seconds later and not 600 seconds later.

When the value for Register Expires is 600 seconds and Proxy Fallback Intvl is 1000 seconds, the backup activates at 1200 seconds.

After successfully re-registering on the primary server, all SIP messages go to the primary server.

How to manage SIP NOTIFY XML-Service support on Cisco SPA512G?

The Cisco SPA500 Series IP Phones support the SIP NOTIFY XML-Service event. Upon receiving a SIP NOTIFY message with an XML service event, the IP phone challenges the NOTIFY with a 401 response if the message does not contain the correct credentials.

The customer must provide the correct credentials using the MD5 digest with the SIP account password for the corresponding line of the IP phone. 

The message body can contain the XML event message. For example:

<CiscoIPPhoneExecute>

<ExecuteItem Priority=”0″ URL=”http://xmlserver.com/event.xml”/&>

</CiscoIPPhoneExecute>.

Authentication:

challenge = MD5( MD5(A1) “:” nonce “:” nc-value “:” cnonce “:” qop-value
“:” MD5(A2) )
where A1 = username “:” realm “:” passwd
and A2 = Method “:” digest-uri

How to configure SIP on Cisco SPA512G?

SIP settings for the Cisco SPA IP phones are configured for the phone in general and for individual extensions.

How to configure basic SIP settings on Cisco SPA512G?

To configure general SIP parameters, go to administrator session > advanced > Voice > SIP. InSIP Parameters, make these changes:

• Maximum forwarding

The number of proxy servers or gateways that can forward the request to the next downstream server. The Max-Forwards value is an integer in the range of 0 to 255 that indicates the remaining number of times the request message is allowed to be forwarded. This count is decremented for each server that forwards the request. The initial value is 70.

• Maximum redirection

Number of times an invitation can be redirected to avoid an infinite loop. The default value is 5. .

• Max Auth

Maximum number of times (from 0 to 255) a request can be challenged. The default value is 2..

• Name of the SIP user agent Agent header

User-Agent header used in outgoing requests. The default value is $VERSION. If empty, the header is not included. Macro expansion from $A to $D corresponding to GPP_A to GPP_D allowed.

• SIP server name

Server header used in responses to incoming replies. The default value is $VERSION..

• SIP Reg User Agent Name

User agent name used in a REGISTER request. If not specified, the SIP user agent name is used for the REGISTER request. 

• SIP acceptance language

Preferred languages for reason phrases, session descriptions or status responses included as message bodies in the response. If blank, the header is not included and the server assumes all languages are acceptable to the client. The default is blank..

• DTMF Retransmission MIME Type

MIME used in a SIP INFO message to signal a DTMF event. This parameter must match that of the service provider. .

Default value is application/dtmf-relay. 

• Hook Flash MIME type

MIME used in a SIPINFO message to signal a hook flash event. .

• Delete last record

If set to yes, it deletes the previous record before re-registering (if the value is different). Default is no.

• Use compact header

If set to yes, the Cisco IP Phone uses compact SIP headers in outgoing SIP messages. If incoming SIP requests contain normal (non-compact) headers, the phone replaces the incoming headers with compact headers..

If set to no, Cisco SPA IP phones use normal SIP headers. If incoming SIP requests contain compact headers, the phones reuse the same compact headers when generating the response, regardless of this setting. Default is no.

• Escape from display name

If set to yes, enclose the configured display name string between a pair of double quotes for outgoing SIP messages. Any occurrence of or in the string is escaped with \ and \ inside the pair of double quotes. The default value is yes.

• Enable SIP-B

If set to yes, enable enterprise SIP calling features (supports Sylantro call flows). See www.broadsoft.com for more information. Default is no. 

• Conversation package

If set to yes, enables support for BroadSoft’s Conversation Pack which allows users to answer or resume a call by clicking a button in an external application. Default is no.

• Wait package

If set to yes, enables support for BroadSoft’s hold package, which allows users to place a call on hold by clicking a button in an external application. Default is no.

• Conference package

If set to yes, enables support for BroadSoft’s Conference Pack which allows users to initiate a conference call by clicking a button in an external application. Default is no.

• Notify conference

If set to yes, Cisco SPA IP phones send a NOTIFICATION with event=conference when initiating a conference call (with BroadSoft conference pack). Default is no.

• RFC 2543 Call Waiting

If set to Yes, Cisco SPA IP phones include the Session Description Protocol (SDP) syntax c=0.0.0.0.0 when sending a new SIP INVITE to a peer to hold the call.

If set to no, Cisco SPA IP phones do not include the c=0.0.0.0.0 syntax in the SDP. With either setting, the phone includes a syntax = send only in the SDP. The default is yes..

• REG CID randomized on reboot

If set to Yes, Cisco SPA IP phones use a different random caller ID for registration after the next software reboot.

If set to no, the phone attempts to use the same caller ID for registration after the next software restart. With either setting, the phone uses a new random caller ID for registration after a power cycle. Default is no. 

• Dial all AVT packets

If set to yes, all audio and video transport tone (AVT) packets (encoded for redundancy) have the marker bit set. If set to no, only the first packet has the marker bit set for each DTMF event..

The default value is yes.

• SIP TCP Port Min

Lowest TCP port number that can be used for SIP sessions.Default value is 5060.

• SIP TCP Port Max

Highest TCP port number that can be used for SIP sessions. The default value is 5080.

• Hold referee when REFERRAL fails

When set to yes, the phone immediately handles NOTIFY sipfrag messages.

• Enable CTI

If set to yes, it enables computer telephony integration (CTI), where a computer can act as a call center handling all types of incoming and outgoing communications, including phone calls, faxes and text messages.

The CTI interface allows a third-party application to control and monitor the status of a Cisco IP Phone and, for example, initiate or answer a call by clicking a mouse on a PC; CTI must be enabled on Cisco IP Phones.

CTI must be enabled on Cisco SPA500 series IP phones in order for a connected Cisco operator console to properly monitor the status of the IP phone line.

Default is no. 

• ID header

Select where the IP phone gets the caller ID from:.

PUT-OFF-RPID-DE.

P-ASSERTED-IDENTITY

REMOTE-PARTY-ID

FROM header

Predetermined in PAID-RPID-FROM.

• SRTP method

The method to use for Secure Real-Time Transport Protocol (SRTP): Method.

x-sipura: legacy SRPT method.

s-descriptor: compatible with RFC-3711 and RFC-4568 The default value is x-sipura.

• Hold target before REFER

Controls whether to hold the call segment with the transfer destination before sending REFER to the transfer receiver when initiating a call transfer with full attention (when the transfer destination has answered). The default value is “no”, where the call segment is not retained..

• Enable SDP Dialog

When enabled and the Notification message body is too large causing fragmentation, the Notification message xml dialog is simplified; the Session Description Protocol (SDP) is not included in the xml content of the dialog.

• Show informationof deviation

Analyzes the divert header information in an incoming SIP invitation and displays it as the caller ID.

• Disable local name to header

The options are No and Yes:.

• If No is selected, no changes are made. The default value is No.
• If Yes is selected, the following occurs:.
  • Disables the display name in “Directory” and “Call History” in the “To” header during an outgoing call.
  • Ignores the CLID in the “UPDATE” message.
  • The redial list is populated according to the PAID 18x or 200 OK header with or without a display name.

Note This field is compatible with firmware version 7.6.2 and later.

How to configure SIP timer in Cisco SPA512G?

All SIP timer values are in seconds. To configure the SIP timer values, go to administrator session > advanced > Voice > SIP. InSIP Timer Values (sec), make these changes:

• SIP T1

RFC-3261 T1 value (RTT estimate). Ranges from 0 to 64 seconds. The default value is 0.5 seconds. .

• SIP T2

RFC-3261 T2 value, the maximum retransmission interval for non-INVITE requests and INVITE responses. It ranges from 0 to 64 seconds. The default value is 4 seconds..

• SIP T4

RFC-3261 T4 value, which is the maximum time a message stays on the network. It ranges from 0 to 64 seconds. The default value is 5 seconds.

• SIP Timer B

RFC-3261 vInvite transaction timeout value. It ranges from 0 to 64 seconds. The default value is 16 seconds.

• SIP Timer F

RFC-3261 vnon INVITE transaction timeout value. Ranges from 0 to 64 seconds. The default value is 16 seconds.

• SIP Timer HRFC-3261

RelINVITE vfinal response timeout value for ACK reception. It ranges from 0 to 64 seconds. The default value is 16 seconds..

• SIP Timer D RFC-3261

Timeout for response retransmissions. Ranges from 0 to 64 seconds. The default value is 16 seconds.

• SIP J timer

FC-3261 Timeout for non-INVITE request retransmissions. Ranges from 0 to 64 seconds. The default value is 16 seconds.

• INVITE expires

The amount of time INVITE is valid. If you enter 0, the Expires header is not included in the request.from 0 to .

19999999999999999999999999999999. The default value is 240 seconds. 

• ReINVITE expires

The ReINVITE request expires header value. If you enter 0, the Expires header is not included in the request.is from 0 to .

199999999999999999999999999999999. The default value is 30.

• Reg Min expires

The minimum registration expiration allowed from the proxy in the Expires header or as a contact header parameter. If the proxy returns a value less than this setting, the lesser of the two values is used. The default value is 1 second..

• Reg Max Expires

Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is greater than this setting, the larger of the two values is used. The default value is 7200 seconds..

• Reg Retry Intvl1

Wait interval before the Cisco IP Phone retries registration after failing during the previous registration. The range is 1 to 268435455. Do not enter 0. The default value is 30 seconds.

• Reg Retry Long Intvl1*

When registration fails with a SIP response code that does not match the Retry Reg response status code (RSC) value (see the following table), the IP phone waits this period of time before retrying.

If this interval is 0, the Cisco IP Phone stops trying. This value must be much greater than the Reg Retry Intvl value. The range is 0 to 268435455. The default value is 1200 seconds..

**Cisco IP phones may use a RETRY AFTER value when received from a SIP proxy server that is too busy to process a request (message 503 Service Unavailable).

If the response message includes a REATTEMPT AFTER header, the phone waits the specified time before REGISTERING again. If there is no REATTEMPT AFTER header, the phone waits for the value specified in the Registration Retry Interval or Registration Retry Long Interval..

• Random Registration Retry Delay

Random delayrandom added to the Register Retry Intvl value when retrying REGISTER after a failure. Minimum and maximum random delay to be added to the short timer. The range is 0 to 268435455. The default value is 0, which disables this function..

• Random long record retry delay

Random delay added to the Register Retry Long Intvl value when retrying REGISTER after a failure. 

Minimum and maximum random delay to be added to the long timer. Random delay interval (in seconds) to add to Register Retry Long Intvl when RE-attempting to REGISTER after a failure. The default value is 0, which disables this function. 

• Reg Retry Intvl Cap

Reg_Retry_Intvl_Cap: maximum value of the exponential delay. The maximum value to limit the exponential rollback retry delay (which starts at Register Retry Intvl and doubles on each retry).

The default value is 0, which disables exponential rollback (i.e., the error retry interval is always in Register Retry Intvl). When this function is enabled, the random register retry delay is added to the exponential backoff delay value. The range is 0 to 268435455..

• Sub Min Expires

Thelower limit of the REGISTER (subscription) value expires returned by the proxy server. The range is 0 to 268435455. The default value is 10 seconds..

• Sub Max Expires

Upper limit of the min-expires REGISTER (subscription) value returned by the proxy server in the Min-Expires header. The range is 0 to 268435455. The default value is 7200 seconds..

• Sub Retry Intvl

The retry interval when the last subscription request fails. The range is 0 to 268435455. The default value is 10 seconds.

How to configure response code management on Cisco SPA512G?

To configure answer status code management, underanswer make these changes:

• SIT1 to SIT4 RSC: SIP answer status code for the appropriate special information tone (SIT). If you set SIT1 RSC to 404, when the user places a call and a failure code of 404 is returned, SIT1 tone is played. Reorder or Busy tone is played by default for all failed answer status codes for SIT 1 RSC to SIT 4 RSC. The default is blank. 

• Try Backup RSC: SIP response code that retries a backup server for the current request. The default value is blank. 

• Retry RSC registration: interval that the device waits before retrying registration after a failed registration. The default value is blank. 

How to configure RTP parameters on Cisco SPA512G?

To configure Real-time Transport Protocol (RTP), go to administrator session > advanced > Voice > SIP. In RTP Parameters, set these fields:

• RTP Port Min: minimum port number for RTP transmission and reception. <RTP Port Min> and <RTP Port Max> define a range containing at least 10 ports of even numbers (twice the number of lines); for example, 100-106. The default value is 16384.
• RTP Port Max: maximum port number for RTP transmission and reception. <RTP Port Min> and <RTP Port Max> must define a range containing at least 10 ports of even numbers (twice the number of lines); for example, 100-106. The default value is 16482.
• RTP packet sizeof the packet in seconds. The range is 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. The default value is 0.030.
• Max RTP ICMP ICMP Err: number of successive ICMP errors allowed when transmitting RTP packets to the peer before the Cisco IP Phone terminates the call. If the value is set to 0 (the default), the Cisco IP Phone ignores the ICMP error limit and disables the feature.
• RTCP Tx Interval: interval for sending Real-time Transport Control Protocol (RTCP) sender reports on an active connection. During an active connection, Cisco SPA IP phones send composite RTCP packets. Each composite RTP packet, except the last one, contains a sender report (SR) and a source description (SDES). The last RTCP packet contains an additional BYE packet. Each SR, except the last one, contains a receiver report (RR); the last SR does not carry RR.

The SDES contains CNAME, NAME and TOOL identifiers:

• • CNAME–User ID@Proxy
  • NAME:Display name (or Anonymous if the user blocks caller ID)
  • HERRAMENT: Vendor/Hardware-platform-software-version (such as Cisco SPA9000-5.2.2(SCb)). 

The NTP timestamp used in the SR is a snapshot of the Cisco IP Phone’s local time, not the time reported by an NTP server. 

If the Cisco IP Phone receives an RR from a peer, it attempts to calculate the round-trip delay and display it as the Call Round Trip Delayin the Information section of the phone’s web UI administration page. The range is 0 to 255 seconds. The default value is 0.

• No UDP checksum: select Yes to allow the Cisco IP Phone to calculate the UDP header checksum for SIP messages. Since this increases the computational load, we recommend the default, no (disabled).
• Symmetric RTP: select Yes to enable symmetric RTP operation. When enabled, it sends RTP packets to the source address and port of the last valid incoming RTP packet received. If disabled (or before the first RTP packet arrives), it sends RTP to the destination as indicated in the incoming SDP. Default is no.
• Statistics in BYE: select Yes to send the P-RTP-Stat header in response to a BYE message. The header contains the RTP statistics on the current call. Default is no. 

The format of the P-RTP-Stat header is: 

P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets received>,PL=<packets lost>,JI=< jitter in ms>,LA=<delay in ms>,DU=<call duration in s>,EN=<encoder;,DE=<decoder&>

How to configure SDP payload types in Cisco SPA512G?

Configured dynamic payloads are used for outbound calls called only when the Cisco IP Phone presents a Session Description Protocol (SDP) offering. For inbound calls with an SDP offering, the phone follows the dynamic payload type assigned to the caller.

Cisco IP phones use the codec names configured in the outbound SDP. For inbound SDP with standard payload types from 0 to 95, the Cisco IP Phone ignores the codec names.

For dynamic payload types, the Cisco IP Phone identifies the codec by the configured codec names (the comparison is case-sensitive).

To configure SDP payload types, go to admin session > advanced > Voice > SIP. Under SDP Payload Types, set these parameters:

• Dynamic AVT payload

Any non-standard data. Both the sender and the receiver must agree on a number.from 96 to 127.

The default value is 101.

• Dynamic INFOREQ payload

Number of used in SIP messaging for the dynamic payload sizing mechanism. This number must match the number configured on the network/other party to enable the use of Dynamic Payload. The best range is 96 to 27 for any type of dynamic payload.

The default value is blank.

• Dynamic payload G726r16

RTP Payload Type Number indicating that the transmitted packet uses the G.726 codec. Other codecs have pre-assigned payload numbers that do not need to be configured, but G.726 does not. The range is 96 to 127.

The default value is 98.

• Dynamic payload G726r24

RTP Payload type number indicating that the transmitted packet uses the G.726-24 codec. It ranges from 96 to 127.

Default value is 97.

• Dynamic payload G726r32

RTP Payload type number indicating that the transmitted packet uses the G726r32 codec. The range is from 0 to 268435455.

The default value is 2.

• Dynamic payload G726r40

RTP Payload Type Number indicating that the transmitted packet uses the G.726-40 codec. The range is from 96 to 27.

Default value is 96.

• Dynamic payload G729b

RTP Payload Type Number indicating that the transmitted packet is using the G729b codec. The range is from 0 to 268435455.

Default value is 99.

• Dynamic EncapRTP payload

EncapRTP dynamic payload type. The range is from 0 to 268435455.

The default value is 112..

• RTP-Start-Loopback Dynamic

RTP-Start-Loopback Dynamic Payload. 

Default value is 113.

• RTP-Start-Loopback Codec

RTP-Start-Loopback CodecRTP-Start-Loopback. Select one of the following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723.

The default value is G711u..

• Code name AVT

AVT code name used in SDP. The default is telephone-event.

• Code name G711u

G.711u code name used in SDP. The default value is Pulse Code Modulation mu-law (PCMU).

• Name of the G.711a code

Code name used in SDP. The default is Pulse Code Modulation A-Law (PCMA).

• Name of the G.726-16 code

Name of the code used in SDP. The default value is G726-16. 

• Name of the G.726-24

Name of the code used in SDP. The default value is G726-24.

• Code name G.726-32

Name of the G.726-32 code used in SDP. The default value is G726-32.

• Name of the G.726-40 code

Name of the G.726-40 code used in SDP. The default value is G726-40. .

• Name of the G729a code

Code name G.729a used in SDP. The default value is G729a.

• Nameof the G.729b code

Name of the G.729b code used in SDP. The default value is G729ab.

• Name of the G723 code

Name of the G.723 code used in SDP. The default value is G723.

• Name of the EncapRTP code

EncapRTP code nameused in SDP. The default is encaprtp. 

How to configure SIP settings for extensions on Cisco SPA512G?

To configure network settings for SIP extensions, go to administrator session > advanced > Voice > Ext_n. In Network Settings, configure the following fields:.

• SIP ToS/DiffServ Value

Value of the Time of Service (ToS)/differentiated services (DiffServ) field in UDP IP packets carrying a SIP message. The default value is 0x68..

• SIP CoS Value [0-7]

Class of Service (CoS) value for SIP messages. The range is 0 to 7. The default value is 3.

• RTP ToS/DiffServ Value

Valueof the ToS/DiffServ field in UDP IP packets carrying RTP data. The default value is 0xb8.

• RTP CoS value [0-7]

CoS value for RTP data. Ranges from 0 to 7. The default value is 6. .

• Network jitter level

The size of the jitter buffer as set by a device. The size of the jitter buffer is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds plus the current RTP frame size), whichever is greater, for all jitter level configurations.

However, the initial value of the jitter buffer size is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum milliseconds.

Select: low (30 ms), medium (40 ms), high (60 ms), very high (80 ms) or extremely high (180 ms). Defaults to high. 

• Adjustment of the jitter buffer

How to adjust the jitter buffer. Select: up and down, up only, downor disable. The default values are above and below.

To configure SIP settings, go to admin session > advanced > Voice > Ext_n. InSIP Settings, configure the following fields:.

• SIP Transport

Select between UDP, TCP or TLS. The default value is UDP..

• SIP port

Port number of the SIP message transmitting and listening port.Default is 5060. 

• Enable SIP 100REL

SIP 100REL extension support for reliable transmission of interim (18x) responses and use of PRACK requests. Select yes to enable. 

Otherwise, select no. Default is no.

• SIP EXT port

The external SIP port number substituted for the actual SIP port in all outgoing SIP messages. If 0 is specified, no SIP port substitution is performed. The default value is blank. The range is from 0 to 65535..

• Auth Resync-Reboot

The Cisco IP Phone authenticates the sender when it receives a NOTIFY message with the following requests:

• • resynchronize
  • reinitialize
  • report
  • reinitialize
  • XML service

Select Yes to enable. Otherwise, select no. The default value is yes..

• SIP Proxy-Require Proxy

SIP proxy for each extension or behavior when an extension sees this user agent header. If this field is set and the proxy does not support it, it responds with the message, not supported. Enter the appropriate header in the field provided. For example, com.nortel.networks.firewall..

• SIP Remote-Party-ID

The Remote-Party-ID header to be used in place of the From header. Select yes to enable. Otherwise, select no. The default value is yes..

• Referor Bye Delay

Time in which the Cisco IP Phone sends BYE to end obsolete call legs after call transfers are completed. Multiple delay settings (reference, reference target, referee, and reference target). Enter the appropriate time period in seconds.Default value is 4.

• Refer-To Target Contact

Indicates the reference destination. Select yes to send the SIP Refer to the contact. Otherwise, select no. Default is no..

• Referee Bye Delay

Delay time for referee bye delay. Enter the appropriate time period in seconds.default value is 0.

• SIP Debug Option

How SIP messages are received or sent from the proxy listening port to the registry. Select:

• • none: no logging. 
  • 1 line: logs the start line only for all messages. 
  • 1 line excl. OPT: logs the start line only for all messages except OPT requests/responses.
• 1 line excl. NTFY: logs the start line only for all messages except NOTIFY requests/responses. 
• 1 line excl. REG: logs the start line only for all messages except REGISTRY requests/responses. 
• 1 line excl. OPT|NTFY|REG: logs the start line only for all messages except OPT, NOTIFY and REGISTER requests/responses. 
• complete: logs all SIP messages in full text. 
• excl. full OPT: logs all SIP messages in full text, except OPT requests/responses.
• excl. full NTFY: logs all SIP messages in full text except NOTIFY requests/responses. 
• excl. full REG: logs all SIP messages in full text except REGISTER requests/responses. 
• excl. full OPT|NTFY|REG: logs all SIP messages in full text except OPT, NOTIFY and REGISTER requests/responses. 

Default value is none.

• Refer Target Bye Delay

Delay time for the reference target bye delay. Enter the appropriate time period in seconds.default value is 0.

• Sticky 183

When enabled, IP telephony ignores additional 180 SIP responses after receiving the first 183 SIP response for an outgoing INVITE. To enable this feature, select yes. Otherwise, select no. Default is no.

• Auth INVITE

When enabled, authorization is required for initial inbound INVITE requests from the SIP proxy. To enable this feature, select yes. Otherwise, select no. Default is no.

• Ntfy Refer On 1xx-To-Inv

When enabled as transferred, the phone sends a NOTIFICATION with Event:Refer to the transferer for any 1xx response returned by the transfer destination, on the transfer call leg. To enable this feature, select yes. 

If set to no, the phone only sends a NOTIFICATION for final responses (200 and above).

• Use Anonymous with RPID

When enabled and the caller blocks their caller ID, the user ID and display name fields of the FROM header are set to anonymous. This parameter is applied only if <SIP Remote-Party-ID> is set to yes; otherwise, it is ignored. 

When disabled, the FROM header display name and user ID are not masked. The Remote-Party-ID header indicates privacy=full when the caller attempts to block your caller ID. 

To enable this feature, select yes. Otherwise, select no. The default value is yes.

• Set G729 appendob

G.729 Annex B (G.729b) which provides silence compression by enabling a voice activity detection (VAD) module. It uses 2-byte silence insertion descriptor (SID) frames transmitted to initiate comfort noise generation (CNG).

If transmission is stopped and the link is muted because no voice is transmitted, the receiving side may assume that the link has been cut.

When inserting comfort noise, analog hiss is digitally simulated during mute to assure the receiver that the link is active and operational. none: do not enable. no: turn on but do not mute the VAD. yes-enable..

• Voice quality report address

The name of the collector that collects SIP PUBLISH event statistics. For example, collector@domain-name-completely-qualified .

([email protected]) or collector@IP–address .

([email protected]). The SIP event package, SIP PUBLISH, allows for the collection and reporting of metrics that measure the quality of VoIP sessions.

Voice call quality information derived from RTCP-XR and SIP call information is transmitted from a user agent in a session to a third party in the SIP PUBLISH method.

RTCP-XR must be configured first (see Configuring RTP parameters). Then ).

enabling RTCP-XR, the call status information is updated in the Voice > Information during an active call. In addition, RTCP-XR packets containing a voice metrics block report are sent with the interval specified in RTCP Tx Interval.

When the call session ends, a SIP PUBLISH with voice metrics information is sent to the collector endpoint. 

This parameter supports a full SIP URI. Examples of valid addresses:.

• • [email protected]
  • [email protected]:5555
  • [email protected]:5656

For example, if extension 1 was configured using the phone profile:

<Voice_Quality_Report_Address_1_ ua=”na”> collector@domain .com

</ emex_quality_report_address_1_> o.

<voice_quality_report_address_1_ ua = “na”>.

[email protected]: 5555

</ iice_quality_report_address_1_> o.

<voice_quality_report_address_1_ ua = “na”>.

[email protected]: 5656

</ emex_quality_report_address_1_>.

How to configure a SIP Proxy Server on Cisco SPA512G?

To configure the SIP proxy and registration parameters:.

Step 1 Go to administrator session > advanced > Voice > Ext_n. 

Step 2 InProxy and registration, configure the following fields:.

• Proxy

SIP proxy server and port number set by the service provider for all outgoing requests. For example: 192.168.2.100:6060. 

The port number is optional. The default is port 5060. The port number is optional.

• Use Outbound Proxy

The outbound proxy (for example, 172.20.2.1:5060; port is optional) or a domain name, such as sip.server.com, provided this name is a fully qualified domain. name. When set to no, Outbound Proxy and Use OB Proxy in Dialog parameters are ignoredThe default is no.

Optionally, the proxy can be configured (Cisco SPA300 or only).

Cisco SPA500 Series) for Survivable Remote Site Telephony (SRST) support. The proxy is configured with an extension that includes a statically configured DNS SRV DNS record or a DNS A record.

The proxy configuration enables failover and failback functionality with a secondary proxy server. For example:

For SRV registration:

sip.server.com:SRV=node1.sip.server.com:5060:p=1:w=50|node

2.sip.server.com:5060:p=2:w= 50

For a record: sip.server.com:A=172.20.2.1,172.20.2.2.2

In both examples, use DNS SRVat no and DNS SRV Auto Prefixis set to no.

• Outbound Proxy

All outbound requests are sent as a first hop. Enter an IP address or domain name.

• Use OB Proxy In Dialog

SIP requests are sent to the outbound proxy within a dialog. This field is ignored if Use outbound proxyis set to noor outbound proxyis blank. To enable this feature, select yes. Otherwise, select no. The default value is yes.

• Registration

Enables periodic logging with the proxy. This parameter is ignored if no proxy is specified. To enable this feature, select yes. 

Otherwise, select no. The default value is yes.

• Make callswithout registration

Enables outgoing calls without a successful (dynamic) registration by the phone. If set to no, dial tone is played only when registration is successful. To enable this feature, select yes. 

Otherwise, select no. Default is no.

• Registration expires

Defines how often the phone renews registration with the proxy. If the proxy responds to a REGISTRATION with a lower expiration value, the phone renews the registration based on that lower value instead of the configured value. .

If the registration fails with an “Expires too short” error response, the phone retries with the value specified in the Min-Expires header of the error..

The range is 0 to 268435455. The default value is 3600 seconds. .

• Ans Call Without Reg

The user does not have to be registered with the proxy to answer calls. To enable this feature, select yes. Otherwise, select no. Default is no. 

• Use DNS SRV

Enable DNS SRV lookup for the proxy and the outbound proxy. To enable this feature, select yes. Otherwise, select no. Default is no.

• Automatic DNS SRV prefix

The phone automatically prefixes the outgoing proxy or proxy name with _sip._udp when performing a DNS SRV lookup on that name. To enable this feature, select yes. Otherwise, select no. Default is no.

• Proxy Fallback Intvl

Sets the delay after which the phone retries from the higher priority proxy (or outbound proxy) after failover to a lower priority server. 

The phone should have the list of primary and backup proxy servers from a DNS SRV record lookup on the server name. It needs to know the priority of the proxy; otherwise, it does not retry.

The range is 0 to 65535. The default value is 3600 seconds.

• Proxy redundancy method

The phone creates an internal list of proxy servers returned in DNS SRV records. 

Select Normal to create a list containing proxies sorted by weight and priority.

Select SRV-based and the phone creates a Normal list, then inspects the port numbers according to the first proxy port listed. When the weight and priority match, the device selects the first port in the list. Default to Normal.

Step 3 Click Submit all changes.

How to configure subscriber information settings in Cisco SPA512G?

To configure subscriber information parameters for each extension, go to administrator session > advanced > Voice > Ext_n. InSubscriber Information, set up the following fields:.

• Display Name

Name that is displayed as the caller ID.

• User ID

Extension number for this line.

• Password

Password for this line. Default is blank (no password required). .

• Use authentication ID

Enables authentication ID and password for SIP authentication. To enable this feature, select yes. Otherwise, select no. Default is no.

• ID of authentication

Authentication ID for SIP authentication. The default is blank.

• Minicertificate

Base64 encoding of a mini-certificate concatenated with the 1024-bit public key of the certificate authority (CA) signing the mini-certificate for all subscribers in the group. The default value is blank..

• SRTP private key

Base64 encoding of the 512-bit private key per subscriber for establishing a secure call. The default value is blank.

• Domain reverse authentication

The IP address for an authentication domain other than the proxy IP address. The default value is blank; the proxy IP address is used as the authentication domain.

The parameter for extension 1 appears as follows in the phone configuration file:.

<Reversed_Auth_Realm_1_ ua=”na”>

</Reversed_Auth_Realm_1_>

How to configure the Cisco SPA512G IP Phone communications protocol?

By default, the phone automatically detects the protocol and the unified communications device.

Cisco SPA500 series IP phones can be used as part of a Cisco Unified Communications system that uses the Smart Phone Control Protocol (SPCP), also known as the Slim Call Control Protocol (SCCP), to manage

a voice network. Or phones can be configured to use Session Initiation Protocol (SIP), an IETF-defined signaling protocol that controls voice communication sessions over an IP network. 

To configure the protocol on a Cisco SPA500 series IP phone, go to administrator session > advanced > Voice > System. In System Configuration under Signaling Protocol , choose SCCP or SIP.

To set the phone to automatically detect the protocol in use on the network to which it is connected, in the Auto SPCP detection field, choose yes.

The phone defaults to SIP unless it detects a Cisco Unified Communications device. When set to no, the phone uses the protocol set in the Signaling Protocol .

How to manage NAT traversal with Cisco SPA512G IP phones?

Network address translation (NAT) allows multiple devices to share a single, public, routable IP address to establish connections over the Internet.

NAT is present in many broadband access devices to translate public and private IP addresses. For VoIP to coexist with NAT, NAT traversal is required. 

Not all service providers offer NAT traversal. If your service provider does not provide NAT traversal, you have several options:

• NAT assignment with the session border controller
• NAT assignment with the SIP-ALG router
• NAT assignment with a static IP address
• NAT assignment with STUN

How to perform NAT mapping with session border controller on Cisco SPA512G?

We recommend that you choose a service provider that supports NAT mapping through a session edge controller. With NAT mapping provided by the service provider, you have more options for selecting a router.

How to perform NAT mapping with SIP-ALG router on Cisco SPA512G?

The NAT mapping can be accomplished by using a router that has a SIP application layer gateway (ALG). By using a SIP-ALG router, you have more options for selecting a service provider. 

How to perform NAT mapping with a static IP address?

You can configure NAT mapping on the phone to ensure interoperability with the service provider. 

• You must have an external (public) IP address that is static. 

• The NAT mechanism used in the router must be symmetric. See Determine whether the router uses symmetric or asymmetric NAT.

Use NAT mapping only if the service provider’s network does not provide session border controller functionality. To configure NAT mapping on the phone:

Step 1 Click Voice > SIP and navigate to NAT support parameters.

Step 2 Set the following parameters to yes:.

• Handle VIA received– Insert VIA received,
• Replace VIA Addr
• Manage VIA rport
• Insert VIA rport
• Send response to Src port

Step 3 Enter the public IP address for the EXT IP 

Step 4 Click on the Ext_n and go to NAT Configuration.

Step 5 Set Enable NAT mapping to yes.

Step 6 (Optional) Set NAT Keep Alive Enable to yes. 

Your service provider may require the phone to send active maintenance NAT messages to keep NAT ports open. Check with your service provider to determine the requirements. .

Step 7 Click Submit all changes.

Step 8 Configure the firewall settings on your router to allow SIP traffic. See SIP Configuration.

How to perform NAT mapping with STUN on Cisco SPA512G?

If the service provider’s network does not provide session edge controller functionality and if the other requirements are met, it is possible to use the session traversal utilities for NAT (STUN) to discover NAT mapping.

The STUN protocol allows applications operating behind a network address translator (NAT) to discover the presence of the network address translator and obtain the assigned (public) IP address (NAT addresses) and the port number that the NAT has assigned for the user datagram.

Protocol (UDP) connections to remote hosts. The protocol requires the assistance of a third-party network server (STUN server) located on the opposite (public) side of the NAT, usually the public Internet.

This option is considered a last resort and should be used only if the other methods are not available. To use STUN.

• The router must use asymmetric NAT. See Determine whether the router uses symmetric or asymmetric NAT.

• A computer running STUN server software is available on the network. You can also use a public STUN server or set up your own STUN server.

Step 1 Click Voice > SIP and navigate to NAT support parameters.

Step 2 Set the following parameters to yes:.

• Handle VIA received– Insert VIA received
• Replace VIA Addr
• Manage VIA rport
• Insert VIA rport
• Send response to Src port
• Enable STUN

Step 3 Enter the IP address for your STUN server in the STUN Server 

Step 4 Click on Ext_n.

Step 5 Set Enable NAT mappingto yes.

Step 6 (Optional) Set NAT Keep Alive Enable to yes. 

Your service provider may require the phone to send active maintenance NAT messages to keep NAT ports open. Check with your service provider to determine the requirements. .

Step 7 Click Submit all changes.

Step 8 Configure the firewall settings on your router to allow SIP traffic. See SIP Configuration.

How to determine whether the router uses symmetric or asymmetric NAT on Cisco SPA512G?

STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from an internal IP address and port to an external routable destination IP address and port.

If another packet is sent from the same source IP address and port to a different destination, a different combination of IP address and port number is used.

This method is restrictive because an external host can send a packet to a particular port on the internal hostonly ifthe internal host first sent a packet from that port to the external host. 

This procedure assumes that a syslog server is configured and ready to receive syslog messages. 

To determine whether the router is using symmetric or asymmetric NAT:

Step 1 Verify that the firewall is not running on your PC. (You can block the syslog port). By default, the syslog port is 514.))

Step 2 Click Voice > System and go to Optional Network Settings. .

Step 3 Enter the IP address for the debug serverand the port number of your syslog server, if the port number is different from the default, 514. You do not need to include the port number if it is the default. 

The address and port number must be accessible from the Cisco IP phone. The port number appears in the name of the output log file. The default output file is syslog.514.log (if the port number was not specified).

Step 4 Set the debug level to 3

Step 5 To capture SIP signaling messages, click the Ext and navigate to SIP Settings. Set the SIP debugging option to Complete. 

Step 6 To collect information about what type of NAT your router uses, click on the SIP and navigate to NAT Support Parameters.

Step 7 Click Voice > SIP and navigate to NAT Support Parameters. .

Step 8 Configure STUN test activation to yes_ .

Step 9 Determine the NAT type by viewing the debug messages in the log file. If the messages indicate that the device is using symmetric NAT, you cannot use STUN.

Step 10 Click Submit all changes.

How to configure VLAN settings on Cisco SPA512G?

If you use a VLAN, the voice packets from your IP phone are tagged with the VLAN ID.

How to configure the Cisco Discovery Protocol (CDP) SPA512G?

The CDP is negotiation-based and determines which VLAN the IP phone resides in. If you are using a Cisco switch, Cisco Discovery Protocol (CDP) is available and enabled by default. CDP:

• Obtains the protocol addresses of neighboring devices and discovers the platform of those devices. 
• Displays information about the interfaces used by your router. 
• It is media and protocol independent.

If you are using a VLAN without CDP, you must enter a VLAN ID for the IP phone. 

How to perform LLDP-MED configuration on Cisco SPA512G?

The Cisco SPA300 and Cisco SPA500 series IP Phones support the Link Layer Discovery Protocol for Media Edge Devices (LLDP-MED) for deployment with Cisco or third-party network connectivity devices that use an automatic Layer 2 discovery mechanism.

The LLDP-MED implementation is in accordance with the IEEE 802.1AB (LLDP) specification dated May 2005 and ANSI TIA-1057 dated April 2006. 

Cisco SPA IP Phones function as LLDP-MED Media End Point Class III devices with LLDP-MED direct links to network connectivity devices, in accordance with the Media End Point Discovery Reference Model and Definition (ANSI TIA-1057 Section 6).

Cisco SPA IP Phones only support the following limited set of TLVs as a Class III LLDP-MED Media Endpoint device:

• TLV chassis
• TLV Port ID Lifetime TLV Port ID
• TLV Port Description
• TLV System Name TLV System TLV Capabilities
• IEEE 802.3 MAC/PHY Configuration/Status TLV (for wired network only)
• LLDP-MED TLV Capabilities
• LLDP-MED Network Policy TLV (for application type = Voice only)
• LLDP-MED Extended Power-Via-MDI TLV (for wired network only)
• LLDP-MED firmware revision
• End of LLDPDU TLV

The outgoing LLDPDU contains all of the above TLVs where applicable. For the incoming LLDPDU, the LLDPDU is discarded if any of the following TLVs are missing. All other TLVs are not validated or ignored.

• TLV ID chassis
• TLV port
• Lifetime TLV
• Capacity TLV
• LLDP-MED LLDP-MED network policy TLV (for application type=Voice only)
• LLDPDU End TLV

The phone sends the shutdown LLDPDU, if applicable. The LLDPDU frame contains the following TLVs:

• TLV chassis
• port ID TLV
• time-to-live TLV
• LLDPDU end TLV

There are some restrictions on the implementation of LLDP-MED on Cisco IP Phones:

• Neighbor information storage and retrieval is not supported.

• SNMP and corresponding MIB are not supported.

• Statistical counter logging and retrieval are not supported.

• There is no full validation of all TLVs; TLVs that do not apply to phones are ignored.

• Protocol state machines, as stated in the standards, are only used as reference.

What is the content of the TLV information?

These sections provide the TLV information.

Chassis ID TLV

For the outgoing LLDPDU, the TLV supports subtype = 5 (Network Address). When the IP address is known, the Chassis ID value is an octet of the INAN address family number followed by the octet string of the IPv4 address used for voice communication.

If the IP address is unknown, the Chassis ID value is 0.0.0.0. The only INAN address family supported is IPv4. Currently, IPv6 address is not supported for the chassis ID. For the incoming LLDPDU, the chassis ID is treated as an opaque value to form the MSAP identifier.

The value is not validated with its subtype. The chassis ID TLV is mandatory as the first TLV. Only one Chassis ID TLV is allowed for outgoing and incoming LLDPDUs.

Port ID TLV

For outgoing LLDPDU, the TLV supports subtype = 3 (MAC address). The 6 octet MAC address for the Ethernet port is used for the port ID value in wired or wireless mode.

For the incoming LLDPDU, the port ID TLV is treated as an opaque value to form the MSAP identifier. The value is not validated with its subtype. The Port ID TLV is mandatory as the second TLV.

Only one Port ID TLV is allowed for outgoing and incoming LLDPDUs..

TLV time-to-live

For the outgoing LLDPDU, the time-to-live TTL value is 180 seconds. This is different from the 120 seconds recommended by the standard.

For the shutdown LLDPDU, the TTL value is always 0. The time-to-live TLV is mandatory as the third TLV. Only one time-to-live TLV is allowed for outgoing and incoming LLDPDUs..

End of LLDPDU TLV

The value is 2 octets, all zero. This TLV is mandatory and only one is allowed for outgoing and incoming LLDPDUs.

Port Description TLV

For outgoing LLDPDU, in port description TLV, the port description value is the same as “port ID TLV” for CDP. The incoming LLDPDU, port description TLV, is ignored and not validated. Only one port description TLV is allowed for incoming and outgoing LLDPDUs.

system name TLV

For outgoing LLDPDU, in the system name TLV, the value is the same as “platform TLV” for CDP. For Cisco SPA525G2, the name is “SPA525G2”. The incoming LLDPDU, the system name TLV, is ignored and not validated. Only one system name TLV is allowed for the .

outgoing and incoming LLDPDUs.

TLV of system capabilities

For outgoing LLDPDU, in the System Capabilities TLV, the bit values for the 2-octet system capabilities field must be set for bit 2 (bridge) and bit 5 (phone) for a phone with a PC port. If the phone does not have a PC port, only bit 5 should be set. .

The same system capability value must be set for the enabled capability field. For the incoming LLDPDU, the system capability TLV is ignored. The TLV is not semantically validated against the MED device type. The system capabilities TLV is mandatory for outgoing LLDPDUs..

Only one system capabilities TLV is allowed.

IEEE 802.3 MAC/PHY Configuration/Status TLV

The TLV is not for automatic negotiation, but for troubleshooting. For incoming LLDPDU, the TLV is ignored and not validated. For outgoing LLDPDU, for TLV, the auto negotiation support/status of the octet value should be:.

• Bit 0: is set to 1 to indicate that the auto negotiation support feature is supported.

• Bit 1: is set to 1 to indicate that the auto negotiation state is enabled.

• Bit 2-7: set to 0.

Bit values for the 2-octet PMD auto-negotiation advertised capacity field should be set to:

• Bit 13: half duplex mode
• 10BASE-T Bit 14: full duplex mode 10BASE-T Bit 11: duplex mode
• 100BASE -Half-duplex mode
• from TX Bit 10: 100BASE-TX full duplex mode
• Bit 15-Unknown

Bits 10, 11, 13 and 14 must be set. 

The value for the 2 octet operational MAU type should be set to reflect the actual operational MAU type: 16:.

• full duplex
• 100BASE-TX 15: half duplex 100BASE-TX
• 11: full duplex
• 10BASE-T 10: half duplex 10BASE-T

For example, in most cases, the phone is set to 100BASE-TX full duplex. The TLV must then be set to 16. The TLV is optional for a wired network and not applicable for a wireless network.

The phone will send this TLV only when in wired mode. When the phone is not configured for auto negotiation, but for a specific speed/duplex, for the outgoing LLDPDU TLV, bit 1 for byte value auto negotiation support/status must be clear (0) to indicate that auto negotiation is disabled.

The 2-octet PMD auto-negotiation advertised capability field must be set to 0x8000 to indicate unknown. The Cisco SPA525G/525G2 allows the administrator to set the switch operating mode to auto-negotiation or to a specific speed/duplexity.

TLV of LLDP-MED capabilities

For the outgoing LLDPDU, the TLV must have device type 3 (endpoint class III) and with the following bits set for the 2-octet capability field:

For the incoming TLV, if the LLDP-MED TLV is not present, the LLDPDU is discarded. The LLDP-MED Capabilities TLV is mandatory and only one is allowed for both outgoing and incoming LLDPDUs. Outgoing and incoming LLDPDUs.

Any other LLDP-MED TLV will be ignored if presented before the LLDP-MED Capabilities TLV.

Network Policy TLV

Outgoing LLDPDU: the telephone will send only a network policy TLV with the application type value set to 1 (voice). Before the VLAN or DSCP is determined, the unknown policy flag (U) is set to 1. application type value.

If the VLAN or DSCP configuration is known, the value is set to 0. When the policy is unknown, all other values are set to 0. Before the VLAN is determined or used, the Tagged Flag (T) is set to 0.

If the tagged VLAN (VLAN ID > 1) is used for the phone, the Tagged Indicator (T) is set to 1. Reserved (X) is always set to 0.

If VLAN is used, the corresponding VLAN ID and L2 priority will be set accordingly. The valid value of VLAN ID is in the range of 1 to 4094. However, VLAN ID=1 (limitation) will never be used. If DSCP is used, the value range from 0 to 63 is set accordingly.

Incoming LLDPDU: several network policy TLVs are allowed for different types of applications. The phone will only support and interpret TLV with application type = 1 (Voice). TLVs for other application types are ignored and will not be validated..

LLLDP-MED Extended Power-Via-MDI TLV

In the TLV for the outgoing LLDPDU, the binary value for Power Type is set to “0 1” to indicate that the power type for the phone is PD Device.

The power supply for the phone is set to “PSE and local” with the binary value “1 1”. The Power Priority is set to binary “0 0 0 0 0” to indicate an unknown priority, while the Power Value is set to the maximum power value according to the phone type:

For incoming LLDPDU, the TLV is ignored and not validated. Only one TLV is allowed on both incoming and outgoing LLDPDUs. The phone will send the TLV only for the wired network. 

The LLDP-MED standard was originally written in the context of Ethernet. Discussion is ongoing.

LLDP-MED for wireless networks. See ANSI-TIA 1057, Annex C, C.3 Applicable TLV for VoWLAN, Table 24. It is recommended that the TLV is not applicable in the context of the wireless network.

This TLV is designed for use in the context of PoE and Ethernet. The TLV, if added, will not provide any value for network management or power policy setting on the switch.

TLV of LLDP-MED inventory management

This TLV is optional for Class III devices. For outbound LLDPDU, we only support firmware revision TLV. The firmware revision value is the firmware version.

For inbound LLDPDU, all TLVs are ignored and not validated. Only one firmware revision TLV is allowed for both outgoing and incoming LLDPDUs.

Final network and QoS policy resolution for the phone

The following sections describe the network policy and QoS for the IP phones.

Special VLANs

VLAN=0, VLAN=1 and VLAN=4095 are treated in the same way as an untagged VLAN. Since the VLAN is untagged, CoS is not applicable..

Default Quality of Service for SIP mode

If no CDP or LLDP-MED network policy exists, the default network policy is used. CoS is based on the specific extension configuration. It is applicable only if manual VLAN is enabled and the manual VLAN ID is not equal to 0, 1 or 4095. ToS is based on the configuration of the specific extension..

Default Quality of Service for SPCP mode

If there is no CDP or LLDP-MED network policy, the default network policy is used. CoS is based on a predefined value of 5.

It is applicable only if manual VLAN is enabled and the manual VLAN ID is not equal to 0, 1 or 4095. T

oS is based on the precedence value of the Unified Communications 500 series StartMediaTransmission message for Cisco SPA525G/525G2. However, ToS is based on the value specified for the specific extension in the web management interface for the Cisco SPA50X IP Phone..

QoS resolution for CDP

If there is a valid CDP network policy:.

• if VLAN = 0, 1 or 4095, the VLAN will not be set or tagged. CoS is not applicable, but DSCP is applicable. ToS is based on the default value as described above. 

• If VLAN > 1 and VLAN < 4095, the VLAN is set accordingly. CoS and ToS are based on the default value as described above. DSCP is applicable.

• For Cisco SPA525G/525G2, when the VLAN is changed, the user sees the voice component updated when the IP address is changed. For Cisco SPA50X, the phone reboots and restarts the quick-start sequence.

QoS resolution for LLDP-MED

If CoS is applicable and if CoS=0, the default value for the specific extension will be used as described above. But the value shown under L2 Priority for TLV for outgoing LLDPDU is based on the value used for extension 1.

If CoS is applicable and if CoS != 0, CoS will be used for all extensions. 

If DSCP (mapped to ToS) is applicable and if DSCP=0, the default value for the specific extension will be used as described above.

But the value shown in DSCP for TLV for outgoing LLDPDU is based on the value used for extension 1. If DSCP is applied and if DSCP != 0, DSCP will be used for all extensions. 

If VLAN > 1 and VLAN < 4095, the VLAN is set accordingly. CoS and ToS are based on the default value as described above. DSCP is applicable..

If there is a valid network policy for the LLDP-MED PDU voice application and if the tagged flag is set, the VLAN, L2 priority (CoS) and DSCP (assigned to ToS) are applicable.

If there is a valid network policy for the LLDP-MED PDU voice application and if the tagged flag is not set, only DPSC (assigned to ToS) applies. 

For Cisco SPA525G/525G2, when the VLAN is changed, the user sees the updated voice component when the IP address is changed. For the Cisco SPA50X, the phone reboots and restarts the quick-start sequence.

Coexistence with CDP

If both CDP and LLDP-MED are enabled, the network policy for the VLAN is determined by the latest policy set or changes with either detection mode. If both LLDP-MED and CDP are enabled, during startup, the phone sends CDP and LLDP-MED PDUs at the same time. 

Inconsistent configuration and behavior of network connectivity devices for CDP and LLDP-MED modes could generate oscillating reboot behavior for the phone due to switching to different VLANs. 

If the VLAN is not set through CDP and LLDP-MED, the VLAN ID that is manually configured is used. If the VLAN ID is not manually configured, no VLAN will be supported. DSCP is used and the network policy is determined by LLDP-MED, if applicable..

Wireless LAN environments

The network policy for the VLAN feature does not support wireless networks. The wireless access point or wireless router must be LLDP-MED enabled as a network connectivity device. The DSCP part for the network policy of the wireless router/AP will be supported if it is enabled..

LLLDP-MED and multiple network devices

If the same application type is used for network policy, but the phones receive different Layer 2 or Layer 3 QoS network policies from multiple network connectivity devices, the last valid network policy is respected.

To ensure a deterministic and consistent network policy, multiple network connectivity devices should not send conflicting network policies for the same type of application. 

LLLDP-MED and IEEE 802.X

The phones do not support IEEE 802.X and will not work in an 802.1X wired environment. However, IEEE 802.1X or spanning tree protocols on network devices could cause a delay in the fast startup response of the switches.

Configuring VLAN settings

To configure VLAN settings, go to administrator session > advanced > Voice > System. Under VLAN Configuration, configure the following parameters:.

• Enable VLAN

Choose Yes to enable VLAN. Choose Noto disable..

• VLAN ID

If you are using a VLAN without Cisco Discovery Protocol (CDP) (VLAN enabled and CDP disabled), enter a VLAN ID for the IP Phone. Note that only voice packets are tagged with the VLAN ID. Do not use 1 for the VLAN ID..

• PC Port VLAN Highest Priority

Choose No limit, or 0–7 (default 0). The highest priority is 7. The priority applies to all frames, labeled and unlabeled. The phone modifies the frame priority only if the incoming frame priority is higher than this value. 

Enable PC port VLAN tagging.

Enable VLAN and priority tagging on the phone data port.

(802.1p/q). This feature facilitates the tagging of the VLAN ID (802.1Q) and priority bits (802.1p) of traffic coming from the PC port of the IP phone. 

Choose Yes to enable the tagging algorithm.Default value is No.

• PC Port VLAN ID

The phone tags all untagged frames coming from the PC. (It does not tag frames with existing tags). The range is 0 to 4095. The default value is 0.

• Enable CDP

Enable CDP only if you are using a switch that has CDP. CDP is based on negotiation and determines which VLAN the IP phone resides in. .

How to customize the standard Cisco SPA512G features?

How to configure the information and display settings of the Cisco SPA512G phone?

The phone’s web user interface allows you to customize settings such as the phone name, background photo, logo, and screensaver.

How to configure the phone name on Cisco SPA512G?

Go to administrator session > advanced > Voice > Phone.

In General, enter the Station Display Name for the phone. This name is displayed on the phone’s LCD GUI in the upper right corner.

How to perform home screen customization on Cisco SPA512G?

You can create a 128 x 48 pixel by 1 bit deep text or logo image to be displayed when the IP phone starts up. (Does not apply to Cisco WIP310 or Cisco SPA501G).

A logo appears during the startup sequence for a brief period after the Cisco logo is displayed.

To set up a custom logo:

Step 1 For Cisco SPA303 and Cisco SPA5XXG, click administrator session > advanced > Voice > Phone.

For Cisco SPA525G or Cisco SPA525G2, click administrator session > advanced > Voice > User.User.

Step 2 To display a text logo, in the Text Logo field enter the text as follows:.

• Up to two lines of text
• Each line must be less than 32 characters
• Insert a newline character (\n) and an escape code (%0a) between the two lines

For example, Super%0aTelecomshows:

Super

Telecom

• Use the +to add spaces for formatting. You can add several +before and after the text to center it.

Step 3 To display an image logo:.

• In the BMP image download URL field, enter the path, for example:

http://192.168.2.244/pictures/image04_128x48.bmp (you can also use a TFTP server)

• Change Select Logoto BMP Image.

Step 4 Click Submit all changes. The phone restarts, retrieves the .bmp file and displays the image the next time it starts up.

Note The supported phone image file types for Cisco SPA500 Series: bitmap format, 1 bit color per pixel, 128 by 48 pixel size.

How to change the wallpaper on Cisco SPA512G?

You can use an image to customize the background of your IP phone’s display.

When BMP image download URL , the phone compares the URL with the previous image URL. (If the URLs are the same, the phone does not download). If the URLs are different, the phone downloads the new image and displays it (provided that the Select Background Image is set to BMP Image).

The phone does not restart after changing the background image URL.

Cisco SPA500 Series.

A background image is displayed while the phone is running. To display a logo during the phone boot sequence.

Step 1 Copy the image to a TFTP or HTTP server that can be accessed from the phone. 

Step 2 Click admin session > advanced > Voice> Phone.

Step 3 Select the background image from the Select Background Image menu:.

• None: does not display a background image.

• BMP image: displays the download URL image .

• Text Logo: displays the text string in the Text Logo field.

Step 4 If you selected None, in Step 3, go to Step 6. If you selected Text logo in Step 3, go to Otherwise, enter the URL of the image file you want in URL of BMP image download.

The URL must include the TFTP or HTTP server name (or IP address), directory and file name, for example:.

tftp://myserver.mydomain.com/images/downloadablepicture.bmp or

http://myserver.mydomain.com/images/downloadablepicture.bmp

If the HTTP refresh timer is set in the server response tothe BMP image download URL, the phone downloads the image from the link and displays it on the IP phone screen. The phone automatically retrieves the image after the specified number of seconds. 

Step 5 If you selected Text Logo, enter a text string in the Text Logo field.

Step 6 Click Submit all changes.

How to configure the screen saver on Cisco SPA512G?

You can configure a screen saver for Cisco SPA500 series IP phones. When the phone is idle for a specified time, it enters screen saver mode. (Users can configure screen savers directly using the Settings button. .

Any button press or off-hook event causes the phone to return to normal mode. If a user password is set, the user must enter it to exit screen saver mode. 

To configure the screen saver:

Cisco SPA5XXG

Step 1 Click administrator session > advanced > Voice> Phone.

Step 2 In the General , in the Enable screen saver field, choose yes to enable it.

Step 3 In the Screen saver wait , enter the number of seconds of idle time that will elapse before the screen saver starts..

Step 4 In the Screen saver icon, choose the display type:.

• A background image.
• The time of the station in the middle of the IP phone screen.
• A moving lock icon. When the phone is locked, the status line displays a scrolling message “Press any key to unlock your phone.”
• A scrolling phone icon.
• The date and time of the station in the center of the IP phone display.
• An energy-saving blank display

Step 5 Click Submit all changes.

How to configure the LCD contrast on Cisco SPA512G?

You can configure the LCD contrast on Cisco SPA500 series IP phones. 

To configure the IP phone display contrast on the phone:

Cisco SPA5XXG

Step 1 Click Admin Login > advanced > User.

Step 2 In LCD, in the LCD Contrast field, enter a numeric value from 1 to 30. The higher the number, the higher the contrast on the IP Phone display. 

Step 3 Click submit all changes.

How to enable calling features on Cisco SPA512G?

This section describes how to enable and disable calling features on the phone.

How to perform secure call activation on Cisco SPA512G?

The phone can encrypt calls to protect them from eavesdroppers. The dial pad codes for encrypting calls are:.

• *16: protects all calls. 
• *17: disable call security that the user enabled by dialing *16. 
• *18: secure an individual call when dialed before or during a call. Use of this star code is redundant if all outgoing calls are already secured by default or by having dialed *16. 

What are the encryption methods for a secure call on Cisco SPA512G?

• SPA IP phones offer two ways to protect a call:

Step 1 Go to administrator session > Advanced > Voice > SIP.

Step 2 In SIP Parameters, configure SRTP Methods. 

Options:.

x-sipura: inherited SRPT method.

s-descriptor:compliant with RFC-3711 and RFC-4568.

• To enable x-sipure type of secure call via mini-certification:

Step 1 Obtain the Generate mini-certificate tool from your service provider. 

Step 2 Go to administrator session > advanced > Voice > Ext_n. 

Step 3 UnderSubscriber Information, enter the minicertificate and the SRTP private key that provide secure encryption of RTP flows between two endpoints on an extension.

Step 4 To enable the secure call service, go to admin session > advanced > Voice > Phone. 

Step 5 Under Additional Services, verify that Secure Calls is set to yes. (This feature can also be set in the User Additional Services tab). 

How to configure secure dial tone on cisco SPA512G?

This tone is played when a call has been successfully switched to secure mode. It should be played only for a short time (less than 30 seconds) and at a reduced level (less than -19 dBm), so that it does not interfere with the conversation. 

To set the tone, go to admin session > advanced > Voice > Regional. In call progress.

Tones, enter the sequence of tones in the Secure call prompt tone field.Default is .

397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2). See Creating scripts for cadences, call progress tones, and tones for syntax information.

How to enable anonymous call blocking and caller ID services on Cisco SPA512G?

To enable anonymous call blocking and caller IDs, go to administrator session > advanced > Voice > User. In Additional Services, under call blocking type, choose yes to enable or no to disable:.

• Block ANC service anonymous calls.
• Block CID Serv-Block outgoing caller ID.

How to enable ACD service on Cisco SPA512G?

Automatic Call Distribution (ACD) is typically used for call centers and handles incoming calls and manages them according to a database of instructions.

Note The ACD feature applies to both Sylantro and Broadsoft. To enable Broadsoft ACD, see Configuring BroadSoft Settings (Cisco SPA300 Series and Cisco SPA500 Series)for Broadsoft ACD support.

To enable ACD:

Step 1 Click admin session > advanced > Voice > User. 

Step 2 In Additional Servicesfrom the ACD Login Service list, choose yesto enable. (The default is no [disabled].)

Step 3In the ACD Ext , choose the extension used to handle ACD calls. Select 1-6 (depending on your phone model).Default is 1.

Step 4 Click SIP and set the SIP B parameter to yes.

Step 5 Click submit all changes.

How to enable callback service on Cisco SPA512G?

Callback forces the phone to repeatedly try a number that received a busy answer. The busy number is tried until the call is placed and the destination phone rings.

To enable the callback service, go to admin session > advanced > Voice > Phone. In Additional Services in the Callback Service field, choose yes to enable.

How to enable call parking and call capture services on Cisco SPA512G?

Call parking allows users to place a call on a line and make it available for another user to pick up. Call pickup allows a user to pick up a phone that is ringing on another user’s phone. 

To enable call parking and call capture, go to admin session > advanced > Voice > Phone. In Additional Services, under Call Feature Type to enable, choose yes to enable or no to disable:

• Call parking service: enables call parking
• Service call pickup: activates call pickup.

How to enable call forwarding and call forwarding services on Cisco SPA512G?

You can transfer or forward a call when the service is enabled.

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 Under Additional Services, under the type of transfer you want to enable, choose yes:.

• Transfer attended: attended call transfer service. The user answers the call before transferring it.
• Blind transfer: blind call transfer service. The user transfers the call without talking to the caller.

You can also enable or disable call forwarding:.

• Cfwd All: Divert all calls.

• Cfwd Busy -Forward calls only if the line is busy.

• Cfwd No Ans –Forward calls only if the line goes unanswered.

Step 3 Click Submit all changes.

How to enable conferencing on Cisco SPA512G?

To enable the user to make conference calls, go to administrator session > advanced > Voice > Phone. In Additional Services in the Conference Service field, choose yes to enable.

How to enable Do Not Disturb on Cisco SPA512G?

You can allow users to enable or disable the Do Not Disturb feature. This feature directs all incoming calls to voicemail or, if voicemail is not set up, plays a message to the caller saying that the user is unavailable.

On the Cisco SPA300 Series and Cisco SPA500 Series IP Phones, users can press the Ignore to forward a ringing call to another destination.

To allow users to use Do Not Disturb (enabled by default), go to admin session > advanced > Voice > Phone. In Additional Services under DND Serv, choose yes to enable. (This feature can also be configured from the User Additional Services tab.)).

How to activate the missed call shortcut on Cisco SPA512G?

IP phones can display a notification that a call has been missed. (Does not apply to Cisco WIP310). 

To enable this notification, go to admin session > advanced > Voice > User. Under Additional Services in the Missed Call Shortcut list, choose yes to enable.

How to enable or disable missed call logging in Cisco SPA512G?

You can disable or enable the missed call log per extension. For example, if you configured a line to monitor another user line, you can disable the missed call log for the monitored line. .

To enable logging, go to admin session > advanced > Voice > User. In Additional Services in the missed call log for extension <number&>field, choose yes to enable.

How to enable the public address (intercom) feature in Cisco SPA512G?

The public address or intercom feature allows two types of public address, single public address and group public address. When paging occurs, the speakerphone of the paged IP phone is automatically activated unless the handset or headset is being used..

A user can directly contact another user by telephone. If the person being paged has configured their phone to accept searches automatically, the phone does not ring; a direct connection between the two phones is established automatically when paging is initiated.

Group paging allows the user to search all of the customer’s Cisco SPA IP phones at once, or search groupsof phones. If the customer’s phone is on an active call while a group page is started, the incoming page is ignored.

The group page is one-way; the paged client IP phones can only hear the originator’s call. 

To enable paging, go to admin session > advanced > Voice > Phone. Under Additional Services in the Paging Services list, choose yes to enable..

To configure a phone to accept pages automatically, go to admin session > advanced > Voice > User. Under Additional Services on the Auto Reply Page , choose yes to enable.

How to configure public address groups on Cisco SPA512G?

You can configure a phone to be a member of a public address group. Then, users can direct pages to specific groups of phones. 

Limitations:

• A phone can be a listening member of no more than two public address groups.

• No more than five public address groups can be configured on a telephone.

To configure a telephone as part of a public address group:.

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 In Multiple Paging Group Parameters, enter the paging commands in the Group Paging Script. The syntax is as follows:.

pggrp=ip-address:port;[name=xxx;]num=xxx;[listen={yes|no}]];.

Where:.

• IP address: multicast IP address of the phone that listens and receives pages.

• port: port to search on; use different ports for each paging group. All phones in the same paging group must use the same port number.

• name (optional): the name of the paging group. In this name, do not use the string pggrp because it is reserved. Using it makes the script not work, as in these examples: 

pggrp=224.168.168.168.168:3141;name=ITGPgGrp;num=800;listen=yes; pggrp=224.168.168.168.168:3141;name=PgGrp;num=800;listen=yes;.

• num: The number that users will dial to access the public address group; must be unique to the group.

• listen: if the phone being configured is a listening member of the page group. A phone can be a listening member of a maximum of two groups. If no value is entered, the default value is not listening as a member of this group.

Step 3 Click Submit all changes.

Example paging group.

This example configures four pagination groups:All, Sales, Supportand Engineering.

Users will press 801 to send alerts to all phones, 802 to send alerts to phones configured as part of the Sales group, 803 to send alerts to phones configured as part of the Support group, and 804 to send alerts to phones configured as part of the Engineering group. 

A phone that is configured with this example is a listening member of the “All” and “Sales” paging groups. That phone will automatically receive pages sent to those two paging groups. For each sales phone, enter the following in the Phone > Multiple Paging Group Parameters > Script :

pggrp=224.123.123.123.121:43210;name=All; num=801;listen=yes; pggrp=224.123.123.123. 121:43211;nombre=Ventas;num=802; escucha=sí; pggrp=224.123.123.121:43212;nombre=Soporte;num=803; pggrp=224.123.123.121:43213;nombre=Ingeniería;num=804;

How to perform service announcement activation on Cisco SPA512G?

Service announcements allow a user to send ad requests to an ad server provided by the customer.

To configure service announcements, go to admin session > advanced > Voice > Phone. Under Ancillary Services, in the Service anc serv , choose yes to enable.

For more information, see Vertical Services Announcement Codes (Cisco SPA300 Series and Cisco SPA500 Series) for detailed configuration. 

How to configure voicemail and message waiting on Cisco SPA512G?

How to configure voicemail on Cisco SPA512G?

This configures the internal or external phone number or URL for the voice mail system. If you are using an external voicemail service, the number must include the digits needed for dialing and any required area codes.

To configure the phone to connect to voicemail:

Step 1 Click admin session > advanced > Voice > Phone Step 2 In General, enter the Voicemail Number. 

Step 3 (Optional) Enter the voicemail subscription interval; the expiration time in seconds, of a subscription to a voicemail server. 

Step 4 Click Submit all changes. The phone restarts.

How to configure the message waiting indicator on Cisco SPA512G?

To enable the indicator at the top of your Cisco SPA300 Series or Cisco SPA500 Series IP Phone to light up when voicemail is left, or on a Cisco WIP310 display a message waiting notification, go to Admin Login > advanced > Voice > Ext_n.

Under Call Feature Settings in the Messages Waiting , choose yes to enable.

How to customize Cisco SPA512G softkeys?

You can customize the softkeys displayed on the phone. The default softkeys (when the phone is idle) are Redial, Directory, Call Forwarding, and Do Not Disturb.

Other softkeys are available during specific call states (e.g., if a call is on hold, the Resume softkey is displayed).

This feature is not available on IP phones using SPCP.

To program softkeys: To program softkeys:

Step 1 Click administrator session > advanced > Voice > Phone.

Step 2 Under Enable programmable softkey, choose yesto enable.

Step 3 Edit the softkeys according to the call status you want the softkey to display. Refer to the table for softkey information. .

In the Softkeys section, each phone state is shown and lists the softkeys that are available to display during that state. Each softkey is separated by a semicolon. Softkeys are displayed in the format: .

softkeyname [position].

where programmableis the key name and the positionis where the key is displayed on the IP phone screen. The positions are numbered; position one is shown at the bottom left of the IP phone display, followed by positions two through four.

Additional positions (more than four) are accessed by pressing the right arrow key on the phone. If a position is not provided for a softkey, the key will floatand appear in the first available empty position on the IP phone’s display.

The following table lists each softkey and the phone status under which the softkey is displayed. You can have a maximum of 16 programmable keys for each call status field.

Step 4 Click submit all changes.

How to define programmable keys in Cisco SPA512G?

The Cisco SPA500 Series IP Phones provide sixteen programmable keys (PSK fields 1 through PSK 16). These keys can be defined using a speed dial script or an XML service script.

To configure the softkeys:

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 Under Enable softkey, choose yesto enable. .

Step 3 In the PSK number field, enter the string for the PSK. Refer to the different PSK types described in the next section. 

Step 4 Click Submit all changes. 

How to configure softkey PSK fields on Cisco SPA512G?

PSKs can be configured as speed dials. Speed dials can be extensions or phone numbers (e.g., like traditional speed dials, where pressing the speed dial will dial a number). 

PSKs can also be configured with speed dials that perform an action defined by a vertical service activation code (also known as star code [*]).

For example, a PSK configured with a speed dial for *67 would place a call on hold. You can also configure PSK to call XML scripts.

To configure a speed dial PSK, enter the following in the PSK number field:.

“fnc=sd;ext=extensionname/starcode@$PROXY;[vid=outboundboundextnum;]nme=name”

where fncis the function of the key (speed dial); extensionname/starcode is the extension to be dialed or the starcode action to be performed; vid is the extension of the calling phone from which the outgoing call is being sent; and name is the name of the speed dial being configured.

The nameis displayed on the softkey on the IP phone display. Cisco recommends a maximum of 8 characters for a Cisco SPA30X or Cisco SPA50X phone and 10 characters for a Cisco SPA525G or Cisco SPA525G2 phone. If more characters are used, the label may be truncated on the IP phone display..

Example speed dial extension.

with softkey This example shows how to configure the Cisco SPA525G or Cisco SPA525G2 phone with a softkey that, when pressed, dials the sales department extension (200).

You want this button to be displayed at the bottom left of the IP phone screen when the phone is idle, when the phone is off-hook, or when the phone is connected on a call.

You want the outgoing call (going to speed dial) to originate from the second extension on the user’s phone, not the main extension.

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 Under Enable programmable softkey, choose yesto enable.

Step 3 In the Programmable Keys section, edit the following:–PSK1: fnc=sd;ext=200@$PROXY;vid=2;nme=.

• List of inactive sales keys: Edit the field to add psk1|1 to the beginning of the string; for example:

psk1|1;em_login;acd_login;acd_logout;avail;unavail; redial;dir;cfwd;dnd;lcr;pickup;gpickup;unpark;em_logout;.

• List of off-hook keys: edit the field to add psk1|1 to the beginning of the string; e.g.:

psk1|1;option;redial;dir;cfwd;dnd;lcr;unpark;pickup;grecogida;.

• List of connected keys: edit the field to add psk1|1 to the string, editing the existing softkey name .|1 to PSK1For example, the original string:

hold|1;endcall|2;conf|3;xfer|4;bxfer;confLx;xferLx;park;phold;flash; becomes:

psk1|1;hold|2;endcall|3;conf|4;xfer;bxfer;confLx;xferLx;park;phold;flash.

Step 4 Click Send all changes. The salesis displayed at the bottom left of the IP phone screen when the phone is idle, when the phone is connected on a call, and when the phone is off-hook.

How to perform toggle configuration for PSK on Cisco SPA512G?

You can configure PSK to toggle or switch between two PSK actions. This is useful when you want a user to be able to switch between two star code actions that have been defined for a PSK.

For example, a PSK could be configured to toggle between enabling and disabling call forwarding using the star code “call forwarding enabled” (*72) and the star code “call forwarding disabled” (*73). 

To configure this type of PSK, enter the following in the “PSK Name” field on the Voice > Phonetab fnc=sd;ext=starcode@$PROXY;nme=name;ext2=starcode@PROXY;nme2 = where fnc=sd is the function of the key (speed dial), starcodeis the star code action to be performed, name is the name of the first action, ext2 is the second star code action to perform and name2 is the name of the second action to perform.

The nameis displayed on the softkey on the IP phone display. Cisco recommends a maximum of 8 characters for a Cisco SPA30X or Cisco SPA50X phone and 10 characters for a Cisco SPA525G or Cisco SPA525G2 phone. If more characters are used, the label may be truncated on the IP phone display..

For example, to configure a call forwarding enabled/disabled PSK that displays in the lower left corner of the IP phone display when the phone is idle:.

Step 1 Click admin session > advanced> Voice > Phone.

Step 2 In Enable programmable softkey programmable, choose yes to enable.

Step 3 In the Softkeys section, edit the following:.

• PSK1: fnc=sd;ext=*72@$PROXY;nme=CFOn;ext2=*73@$PROXY;nme2= CFOff;

• Inactive key list: edit the field to add psk1|1 to the beginning of the string; e.g.:

psk1|1;em_login;acd_login;acd_logout;avail;unavail; redial;dir;cfwd;dnd;lcr;pickup;gpickup;unpark;em_logout;.

Step 4 Click Submit all changes.

How to configure PSK to call XML scripts on Cisco SPA512G?

To configure an XML script, enter the following in the PSK field: fnc=xml;url=http://scriptURL.xml;nme=scriptname.

where fncis the key function (an XML script) , scriptURL. xml is the URL where the script is located and scriptname is the name of the script..

The scriptis displayed on the softkey on the IP phone display. Cisco recommends a maximum of

8 characters for a Cisco SPA300 Series or Cisco SPA500 Series phone and 10 characters for a Cisco SPA525G or Cisco SPA525G2 phone. If more characters are used, the label may be truncated on the IP phone display..

You can use macro variables in XML URLs. The following macro variables are supported: .

• User ID: UID1, UID2
• DisplayName: DISPLAYNAME1, DISPLAYNAME2
• Authentication ID: AUTHID1, AUTHID2
• Ringer tone proxy settings: PROXY1, PROXY2
• MAC address: MA
• Product name: PN
• Product serial number: PSN
• Serial number: SERIAL_NUMBER Step 5 Click on Submit all changes.

How to configure physical standby button and PSK binding on Cisco SPA512G?

You can configure the physical hold button on the phone to perform a star code action that has been configured as a PSK.

For example, with some call control systems, placing a call on hold or resuming a call on hold requires the phone to send a star code to the server.

With those systems, the phone’s physical hold button cannot be used to place a call on hold or resume a call on hold, because a star code is not sent to the server.

You can configure a PSK to perform a hold/resume call action, then configure the physical hold button on the phone to perform that action when pressed. This is done by adding the holdkey=yesto the softkey..

For example:

Step 1 Click administrator session > advanced > Voice > Phone.

Step 2 In Enable programmable softkey programmable, choose yesto enable.

Step 3 In the Softkeys section, edit the following:.

• PSK1: fnc=sd;ext=*67@$PROXY;nme=hold;ext2=*68@$PROXY;nme2= resume;holdkey=yes

• • List of connected keys: edit the field to add psk1|1to the string, editing the existing softkey name . |1 to PSK1For example, the original string:

endcall|1;conf|2;xfer|3;bxfer;confLx;xferLx;park; fhold;flash; becomes:.

psk|1;endcall|2;conf|3;xfer;bxfer;confLx;xferLx;park;phold;flash.

Step 4 Click Send all changes..

How to configure ring tones on Cisco SPA512G?

You can define up to 12 ring tones for a Cisco SPA500 series IP phone. In addition to these 12 ring tones, 4 user-configurable ring tones can be used in place of some of the default ring tones.

See Appendix A, “Ring Tone (Cisco SPA300 Series and Cisco SPA500 Series),”for more information on ring tones.

You can set:

• The default ringtone for the extension
• Specific ringtones assigned to individual callers in the personal directory. These override the default ring tone.

To set ringtones:

Step 1 Click admin session > advanced > Voice > Phone and scroll to the Tone .

Configure the characteristics of each ringtone using a ringtone script. Specify: .

• Name (n): ringtone name, such as Classic, Simple or
• office-wave (w):
• (Not compatible with Cisco SPA300 Series or Cisco SPA5XXG.)
• (c): 1, 2, 3, 4 or 5 (Not compatible with Cisco SPA300 Series or Cisco SPA5XXG).

Step 2 Click Submit All Changes. .

You can also configure four additional ring tones to replace up to four of the default standard ring tones. The following user-configurable ring tones are available:

To set up/provision these ringtones:

Step 1 Click Admin Login > advanced > Voice > Phone and scroll to the Ringer tone .

Step 2 In the Ringtone section, modify the n and win four of the 12 ringing fields (Ring1 to Ring12). Set the n in the label of the ringtone you want the GUI to display. Set the w parameter wshown in the table above..

For example, to replace the ringtone in Ring1 with the Warble ringtone, change the value of the Ring1 field to n=warble;w=7;c=1 or set it as follows in the phone configuration file:.

• Cisco SPA300 Series or Cisco SPA5XXG

< Ring1 ua=”na”>n=gorjeo;w=7;c=1</Ring1>

• (Cisco SPA525G or Cisco SPA525G2)

<Ring1 ua=”na”>n=gorjeo;w=file://Warble.raw ;c=1</Ring1>.

Step 3 Click Send All Changes.

You can also download one of the two available ringtones (user ringtone 1 or 2) using TFTP:.

http://phone_ip_addr/ringtone1?[url]

The tftpsyntax of [url] is ://host[:port ]/path. .

• The default host is the TFTP host.

• Port is optional. The default port is 69.

• The link is case-sensitive. 

On IP phones, ringtones downloaded by the user are labeled as User 1 and User 2 in the Default Ring options.

In the phone’s ringtone menu, the User 1 and User 2 options are replaced with the corresponding ringtone name. Not installed appears if the user ring tone slots are not used. 

For the User 1 and User 2 ring tone, the cadence is set with the on time equal to the duration of the ring tone file and the off time equal to four seconds. The total ringing duration is fixed at 60 seconds.

The user ringtone names displayed on the IP phone screen are derived from the ringtone file header file. 

It is not necessary to reboot the phone after downloading a ringtone. 

To remove user 1’s ringtone from the phone, configure the route to remove it, as follows:

http://phone_ip_addr/ringtone1?/delete

How to create and load ringtones using the ringtone utility on Cisco SPA512G?

To convert a file for use as a ring tone, use Ring Tone Utility, available at:.

https://supportforums.cisco.com/ docs/DOC-9944

You must have a .wav file less than 8 seconds long saved on your computer. You can also use a sound editor to create the file with the following restrictions:.

• 16-bit mono PCM
• 8000 samples per second
• Less than 6000 ms in duration

To create a ringtone and load it into a phone:.

Step 1 Open the Ringtone Utility.

Step 2 Enter the IP address of the phone.

Step 3 Click Examine and navigate to the directory on your computer where the source .wav file is stored. Select the wav file and click Open.

Step 4 Click Load Source File.

Step 5 Enter a name for the ringtone. This name will appear on the phone screen. You choose the file name later.

Step 6 Enter the target. You can have up to two custom ringtones loaded into the phone.

Step 7 (Optional) Click Previewto preview the ringtone. Click Options to change the start or end positions, or to compress or stretch the audio.

Step 8 Click Load to Phone to load the ringtone to the phone. Do Acceptwhen the success status message appears.

Step 9 Close the open Ring Tone utility windows.

To create a ringtone and save it to a file:

Step 1 Open the Ring Tone Utility.

Step 2 Enter the IP address of the user’s phone or press Skipto create the ringtone and save it as a file.

Step 3 Click Browseand navigate to the directory on your computer where the source wav file is stored. Select the wav file and click Open..

Step 4 Click Load.

Step 5 Enter a name for the ringtone. This name will appear on the IP phone’s display. You choose the file name later.

Step 6 (Optional) Click Previewto preview the ringtone. Click Options to change the start or end positions, or to compress or stretch the audio..

Step 7 Click Save As to save the file to your computer. Enter the file name and press Save..

Step 8 Close the open windows of the Ring Tone utility..

To remove a ringtone from a phone:

Step 1 Open the ringtone utility.
Step 2 Enter the IP address of the phone.
Step 3 Click the Delete button next to the ringtone you want to delete.
Step 4 Click OK.
Step 5 Close the Ringtone Utility windows.

How to assign a ringtone to an extension on Cisco SPA512G?

To assign a ring tone to an extension:.

Step 1 Click admin session > advanced > Voice > ExtExt <number > tab.

Step 2 In Call Feature Settingson the Default ringer, choose from the following:.

• No ringer
• 1 to 12
• User 1
• User 2

Step 3 Click Submit all changes.

How to configure audio settings on Cisco SPA512G?

You can configure the default audio settings for the phone. The user can modify the volume settings by pressing the volume control button on the phone and then pressing the SaveprogrammableWIP310 button. .

To configure the audio volume settings:.

Step 1 Click admin session > advanced > Voice > User User

Step 2 In the Audio Volume section, set a volume level between 1 and 10, with 1 being the lowest level:

• Buzzer Volume

Set the ringer volume. 

• Speaker volume

Sets the volume of the full duplex speaker. 

• Headphone volume

Sets the earpiece volume. 

• Headphone volume

Sets the headphone volume.

• Phone version

Phone version: change the phone version manually.

Automatic: phone automatically sets phone version according to hardware version and model.(Default) Original: phone set to version 2 and below.

V3: the handset is set to version 3.

• Deep bass

Set a standard tone or an enhanced bass tone.

• Enable speakerphone

Enable or disable the speaker. If the parameter is set to yes (the default setting), the speakerphone is enabled. If the parameter is set to no, the speakerphone is disabled and pressing the Speaker button on the phone sends audio to the phone’s earpiece instead of the speakerphone..

• Enable Mute

Allows you to enable or disable the Mute button.

The default value is Yes.

If the parameter is set to No, the user cannot mute the audio..

Note This field is compatible with firmware version 7.6.2 and later.

Step 3 Click submit all changes.

How to configure audio input on Cisco SPA512G?

The default value in the headset, headphones, or speakerphone settings is zero, which indicates that the volume is set to a basic level. This does not mean that the sound is turned off; it is set to a level that the average person can hear in a normal office environment. 

To amplify or reduce the sound level, go to admin session > advanced > Voice > Phone. In Audio Input Gain (dB), choose the item to configure:.

• A positive value increases the amplification (the sound is louder).
• A negative value decreases the amplification (the sound is lower).
• Set a value that is high enough to hear clearly without producing echo (an indication that the input gain is too high)

How to enable SMS messaging on Cisco SPA512G?

Cisco SPA IP phones can receive and display text messages via SIP (RFC-3428).

When this feature is enabled, the IP phone display shows messages up to 255 characters in length. The message appears on the IP phone display along with the date and time. 

Service providers may use text messages to:.

• Send billing information, call minutes consumed, minutes available.

• Include additional text with a call to facilitate call processing.

To enable receiving text messages on Cisco SPA500 series phones:

Step 1 Click admin session > advanced > Voice > User.

Step 2 In Additional Servicesin the Text Message , choose yesto enable.

Step 3 (optional) To enable receiving text messages from a third party directly without the involvement of a proxy, in the Message , choose yes to enable.

Step 4 Click submit all changes.

How to activate and configure the Cisco SPA512G phone’s web server?

The web server allows administrators and users to log in to the phone through a web user interface on the phone. Administrators and users have different privileges and see different options for the phone depending on their role.

How to configure the web server from the Cisco SPA512G phone web interface?

To enable the web server:

Step 1 Click administrator session > advanced > System.

Step 2 In the System Configuration Enable Web Server section, verify that the parameter is set to yesto enable the web management server. 

Step 3 In the Web server port , enter the port to access the web server. The default value is port 80..

Step 4 In the Enable web administrator access , you can enable or disable local access to the web administrator session of the phone’s web UI administrator.

The default is yes (enabled). (For Cisco SPA301 and Cisco SPA501G, you can configure Cisco SPA512G using the IVR. See the section “Using IVR on IP phones without displays” on page 1-14).) 

Step 5 In the Admin Passwd, enter a password if you want system administrator to log in to the phone’s web user interface with a password. The password prompt appears when an administrator clicks administrator login. The maximum password length is 32 characters.

Step 6 In the User Password , enter a password if you want users to log in to the phone’s web user interface with a password. The password prompt appears when users click User Login. The maximum password length is 32 characters.

Step 7 Check if the entry is provided in the User Password in the previous step. 

Otherwise, enter a password in the User Web Password if you want users to log in to the phone’s web user interface with a password and no password for the LCD GUI. 

The password prompt appears when users click User Login. The maximum password length is 32 characters.

Step 8 Click on Submit all changes.

To enable the phone’s web user interface from the Phone:

Step 1 Press menu. 

Step 2 Select Network and enable web server.

Step 3 Select Edit.

Step 4 Press y/n to toggle the selection to Yesand enable.

Step 5 Click OK > Save.

The WBPN device (a dongle device) with an integrated web configuration GUI and Windows setup wizard serves as a wireless access point and enables Wi-Fi capability on Cisco SPA30X, SPA50X, and SPA51X phones. .

The Windows WBPN wizard uses a special discovery protocol to communicate with the web configuration GUI (instead of the complicated HTTP protocol) so that SPA phones implement the same interface to configure WBPN.

WBPN has a default IP address, 192.168.1.254. The Windows WBPN wizard uses the default IP address to communicate with the web configuration GUI. (Discovery Protocol packets carry the default IP address as the source IP.)

Before the phone initializes the WBPN module, it obtains a valid IP address in any of the following ways:.

• DHCP
• Static IP IP
• Backup IP
• Selecting the WBPN configuration menu on the phone The configuration module performs the following:
• Discover the WBPN module
• Obtain the configuration
• Scan for Wi-Fi access points
• Set the WBPN configuration
• Execute the WPS procedure in WBPN from the phone
• Get WBPN status

How to configure the WBPN device on Cisco SPA512G?

While pairing the SPA phone with WBPN, connect the WBPN Ethernet cable directly to the Ethernet interface of the SPA phone. After the SPA phone and WBPN start up, configure the WBPN device (from the phone screen) by following these steps: 

Step 1 Press the Configure to access the SPA phone’s LCD menu.

Step 2 Scroll to Wi-Fi Setupand pressSelect.

Step 3 Enter the user IDand password.

Note

• If the WBPN is new, the user IDand passwordare Cisco. Otherwise, enter the user ID and password modified by the user in WBPN webgui.
• A message is displayed on the LCD screen if the user ID or password is incorrect. After a successful login, users do not need to re-enter the user ID and password each time to access the Wi-Fi setupmenu
• If the phone has an IP address and there is no WBPN connected to the IP phone, then:
• The message “No Wi-Fi device found” appears.
• If the phone does not have an IP address:
• The message “Wait” appears (timeout is 15 seconds). The phone stops the DHCP detection process and a temporary IP address 192.168.1.235 is assigned.
• If the phone cannot connect to the WBPN, make sure that the WBPN device is properly initialized. Configure the IP phone with static IP/netmask/gateway 192.168.1.235/255.255.255.255.0/192.168.1.1 and then try to connect to the WBPN again.

Step 4 Select Wireless Profilethe menu.

There are three functions available:

-Edit: to edit the wireless profile.

–Scan: WBPN searches for wireless applications and displays the Wireless Profile List menu.

–Cancel: returns to the Wi-Fi settingsmenu.

Step 5 Select SSIDand press the Editsoftkeyto edit the detailed wi-fi settings.

Three menu options are displayed:.

–Basic configuration: SSID and band.

–Security mode: edit security mode.

–Encryption settings: , key format and key value.

edit encryption edit-environment.

• By selecting the Sendkey, the current settings stored in the phone are sent to the WBPN device. Then, the device reboots and the phone restarts within 4 seconds.

• Encryption configuration is not shown if encryption is not configured for the WBPN device.

Step 6 (Optional) Select Wi-Fi Configuration > Wireless Profile > Scanning Softkeys.

Step 7(Optional) Select Wi-Fi Setup > Wi-Fi Protected to configure WPBN via WPS. (Follow the prompts on the phone’s LCD screen to complete the Wi-Fi setup via WPS.).

How to perform wireless status verification on Cisco SPA512G?

To verify the current Wi-Fi status of the WBPN device, go to Wi-Fi Settings>Wireless Status.

Note Each time the user accesses the Wireless Status menu. The LCD screen of the phone displays the last status after several seconds.

How to configure LDAP on Cisco SPA512G?

The Cisco SPA500 series IP phones support Lightweight Directory Access Protocol (LDAP) v3.

LDAP corporate directory lookup allows a user to search a specific LDAP directory for a name, phone number, or both. LDAP-based directories such as Microsoft Active Directory 2003 and OpenLDAP-based databases are supported. 

Users access LDAP from the Directory on their IP phone. There is a limit of 20 records returned from an LDAP lookup..

The instructions in this section assume that you have the following equipment and services:.

• An LDAP server, such as OpenLDAP or Microsoft Active Directory Server 2003
• A Cisco SPA300 Series or Cisco SPA500 Series IP Phone with firmware version 6.1.3a or higher To prepare the LDAP Corporate Directory Search:

Step 1 Click administrator session > advanced > System.

Step 2 In the Optional Network Configuration Main DNS section, enter the IP address of the DNS server. (Only required if you are using Active Directory with authentication set to MD5.).

Step 3 In the Optional Network Settings Domain section, enter the LDAP domain. (Only required if you are using Active Directory with authentication set to MD5.).

Some sites may not implement DNS internally and instead use Active Directory 2003. In this case, it is not necessary to enter a primary DNS address and LDAP domain. However, with Active Directory 2003, the authentication method is restricted to Simple.

Step 4 Click the Phone tab.

Step 5 In LDAP, in the Enable LDAP Directory field Yes to enable LDAP and make the name defined in LDAP Corp Dir Name appear in the phone directory.

Step 6 Set the values for the fields in the following table and click Submit All Changes. 

• LDAP Corp Dir Name

Enter a free-format text name, such as Corporate Directory.

• LDAP Server

Enter a fully qualified domain name or IP address of the LDAP server, in the format nnn.nnn.nnn.nnn.nnn.nnn.

Enter the hostname of the LDAP server if the MD5 authentication method is used..

• LDAP authentication method

Select the authentication method required by the LDAP server:

None: no authentication is used between client and server. None: no authentication is used between client and server.

Simple: the client sends its full domain name and password to the LDAP server. It could create security problems.

Digest-MD5: the LDAP server sends authentication options and a token to the client. The client returns an encrypted response, which the server decrypts and verifies.

• LDAP Client DN

Enter the domain components of the distinguished name [dc] ; for example: dc=cv2bu,dc=com.

If you use the default Active Directory schema.

(Name(cn)->Users->Domain), example of the client DN: cn=”David Lee”,dc=users,dc=cv2bu ,dc=com

• LDAP user

Enter the username for a credentialed user on the LDAP server.

• LDAP Password

Enter the password for the LDAP user name.

• LDAP search base

Specify a starting point in the directory tree to search from. Separate the components of the [dc] domain with a comma. For example: dc=cv2bu,dc=com.

• LDAP Last Name Filter

Define the search for last name [sn], known as surname in some parts of the world. For example, sn:(sn=*$VALUE*). This looks for the text string anywhere at the beginning, middle, or end of a name..

You must enter a value in the last name and first name fields for the LDAP corporate directory option to display on the phone. If both fields are empty, the directory is not displayed.

• LDAP name filter

Define the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This looks for the text string anywhere at the beginning, middle, or end of a name..

You must enter a value in the last name and first name fields for the LDAP corporate directory option to display on the phone. If both fields are empty, the directory is not displayed.

• LDDAP Search Element 3

Enter a custom search element. It can be blank if it is not required. .

• LDAP element 3 filter

Enter a custom filter for the searched element. It can be blank if it is not required. .

• LDAP search element 4

Enter a custom search element. It can be blank if it is not required.

• LDAP Element 4 Filter

Enter a custom filter for the searched element. It can be blank if it is not required. .

• LDAP Display Attrs

Enter the LDAP results display format on the phone where: .

• • a-Attribute name
  • cn-Common name
  • sn-Surname (last name)
  • phoneNumber-Phone number
  • n-Display name

For example, n=Phonemakes Phone:be displayed in front of the phone number of the result of an LDAP query when the programmable detail button is pressed. t-type.

When t=p, tis of type phone number and the retrieved number can be dialed. Only one number can be dialed. If two numbers are defined as dialable, only the first number is used. For example, a=ipPhone, t=p; a=mobile, t=p;.

In this example, only the can be dialed.ipPhone number and the cell phone number is ignored.

• p: phone number

When pis mapped to a type attribute, example t=p, the retrieved number can be dialed.

• LDAP number assignment

With LDAP number assignment, you can manipulate the number retrieved from the LDAP server. For example, you can add 9 to the number if your dial plan requires the user to enter 9 before dialing. 

Add the prefix 9 by adding (<:9>xx.>)to the LDAP Number Mapping field. For example, 555 1212 will become 9555 1212. It can be left blank if it is not needed..

If you do not manipulate the number in this way, a user can use the Edit dialing to edit the number before dialing.

For more information on LDAP, including troubleshooting information, see the Configuring LDAP Directory Lookup on IP SIP SPA Phones Application Note from .

http://www.cisco.com/web/partners/sell/smb/products/voice_and_conferencing.html#~vc_technical_resources (Partner login required).

How to set up the BroadSoft directory?

BroadSoft’s directory service allows users to search and view their personal, group or business contacts. This application feature uses BroadSoft’s Extended Services Interface (XSI). .

To configure BroadSoft’s directory service:.

Step 1 Click administrator session > advanced > Voice > Phone.

Step 2 In Broadsoft Configuration, configure the following:.

• Enable directory: set it to yes.
• XSI host server: enter the server name; for example, xsp.xdp.com.
• Directory Name: Name of the directory. Appears on the user’s phone as a directory option (e.g., John’s personal directory).
• Directory type: select the BroadSoft directory type:
• Company (default): allow users to search by last name, first name, user or group ID, phone number, extension, department, or email address.
• Group: allows users to search by last name, first name, user ID, phone number, extension, department, or email address.
• Personal: allows users to search by last name, first name or phone number.
• Directory user ID: BroadSoft user ID of the phone user; for example, [email protected].
• Directory Password: alphanumeric password associated with the User ID.

For enhanced security, the SPA phone firmware sets access restrictions on the host server and directory name entry fields.

Step 3 Click Submit all changes.

How to configure the personal address book in Cisco SPA512G?

Users can load a preconfigured personal address book xml file on the phone. While loading the address book information into the phone, provide complete address book entries.

The phone deletes the previous personal address book entries and populates the new entries from the address book xml file. 

After the phone reports the personal address book or call history data to the provisioning server, users can restore the data to the phone after factory reset.

Example of personal address book xml format (for SPA525G, SPA300 series phones and SPA500 series phones use a subset):

<flat-profile>

<paddrbook>

<entry>

<name>Abc Test</name>

<homePhone>4081111234</homePhone>

<workPhone>4082221234</workPhone>

<mobilePhone>4083331234</mobilePhone>

<ringToneID>1</ringToneID>

</entry>

<entry>

<name>Def Test</name>

<homePhone>4081001234</homePhone>

<ringToneID>1</ringToneID>

</entry>

</paddrbook>

</flat-profile>where: name: contact’s name (maximum size: 64; entry type: Any). homePhone: contact’s home phone number (maximum size: 128; entry type: Any). workPhone: contact’s work phone number (maximum size: 128; entry type: Any). 

mobilePhone: contact’s cell phone number (maximum size: 128; input type: Any).

ringToneID: ringtone for the contact, selection of available ringtones (input type: numeric, 0-12).

For information on how to back up the personal address book or call history, refer to SPA Provisioning Guide.

How to synchronize Do Not Disturb and Call Forwarding per line (applicable to BroadSoft) on Cisco SPA512G?

Enabling Do Not Disturb (DND) and Call Forwarding (CFWD) synchronization allows the phone to synchronize with the call server (for example, the BroadSoft server) so that if Do Not Disturb or Call Forwarding settings are changed on the phone, changes are also made on the server; if changes are made on the server, they are propagated to the phone. You can enable DND/CFWD per extension..

This feature is disabled by default.

Limitations:

• Cisco SPA301 or Cisco SPA501G: softkey and phone menu settings are not available.
• Cisco SPA509: lines 9 to 12 cannot be configured using the softkeys or menu settings. 

How to configure DND and CFWD synchronization on Cisco SPA512G?

To enable synchronization:

Step 1 Click admin session > advanced > Speech.

Step 2 Click on the Ext n tab.

Step 3In Settingscall function on the Key Synchronization , choose yes to enable DND/CFWD. .

Step 4 In SIP, enable the relevant event packet (Talk Packet, Standby Packet and Conference Packet).

Step 5 Click Submit all changes.

How to configure DND and CFWD synchronization by using the configuration file in Cisco SPA512G?

You can also configure BroadSoft DND and CFWD per line by modifying your configuration. For example, to configure this feature on line 1, add the following line to the configuration file: <Device_Feature_Sync_1_ ua=”na”>Yes</Device_Feature_Sync_1_>.

How to configure Broadsoft ACD support on Cisco SPA512G?

To support basic Broadsoft Automatic Call Distribution (ACD), enable the Broadsoft ACD Option . This option is available for each extension in Call Feature Configuration..

The supported values for this option are Yes and No (default).

If you set Broadsoft ACDto Yes, the phone sends a subscription message in accordance with Broadsoft’s specification.

If you set Broadsoft ACDto No, the phone can still send a subscription message because another feature is using ACD, but the phone ignores any Broadsoft server notification messages related to .

ACD.

Limitations:

• Cisco SPA301 or Cisco SPA501G-ACD is not supported. ACD status and login keys are not visible.

• Cisco SPA509: lines 9 to 12 cannot be used as ACD agents, as the lines cannot be selected for login/logout and agent status. 

To enable BroadSoft ACD support, go to Admin Login > advanced > Voice > Ext n. In Call Feature Setup, in the Broadsoft ACD , choose Yes to enable BroadSoft ACD support..

You can also configure BroadSoft ACD support by adding the following line to your configuration file to configure this feature on line 1:.

<Broadsoft_ACD_1_ ua=”na”>Yes</Broadsoft_ACD_1_>.

How to configure XML services for Cisco SPA512G?

The Cisco SPA500 series IP phones provide support for XML services, such as an XML directory service or other XML applications. 

For the Cisco SPA512G, the supported services are:

• CiscoIPPhoneMenu
• CiscoIPPhoneText
• CiscoIPPhoneInput
• CiscoIPPhoneDirectory
• CiscoIPPhoneIconMenu
• CiscoIPPhoneStatus
• CiscoIPPhoneExecute
• : CallHistory
• Cisco SPA5XXG
• Key:Headset
• EditDial:n

For more information on how to use Cisco XML, see the Cisco Unified IP Phone Services Application Development Notes.

How to manage the XML directory service for Cisco SPA512G?

When authentication is required for XML URLs, the XML Username and XML Password parameters are used.

The “XML UserName” parameter in the XML URL is replaced with $XML UserName. 

For example:

if the XML UserName parameter is set to “cisco” and the XML directory service URL is http://www.sipurash.compath? username=$XML_User_Name, when the request is sent, the URL will be http: //www.sipurash.com/path?username=cisco.

How to configure XML Services XML applications for Cisco SPA512G?

When authentication is required for CGI/Execute URL via publishing from an external application (Example: web application) to SPA phones, the “CISCO XML EXE Auth Mode” parameter is used in 3 different scenarios:.

• Trusted: no authentication is performed (local user password is set or not). This is the default option.
• Local credential: authentication based on implicit authentication using the local user password, if the local user password is configured. Otherwise, there is no authentication.
• Remote credential: based authentication on implicit authentication using the remote username/password as set in the XML application on the web page (to access the XML application server).

What are the Cisco SPA512G macro variables?

You can use macro variables in XML URLs. The following macro variables are supported:.

• User ID: UID1, UID2 to UIDn
• DisplayName: DISPLAYNAME1, DISPLAYNAME2 to DISPLAYNAMEn
• Authentication ID: AUTHID1, AUTHID2 to AUTHIDn
• Proxy: PROXY1, PROXY2 to PROXYn
• MAC address using lowercase hexadecimal digits: MA
• Product name: PN
• Product Serial Number -Number
• serial PSN-SERIAL_NUMBER

The form $$ expands to a single character $. .

• A to P

Replaced by general purpose parameters GPP_A to GPP_P.

SA to SD

Replaced by special purpose parameters GPP_SA to GPP_SD. These parameters contain keys or passwords used in provisioning. .

Note $SA to $SD are recognized as arguments to the optional resynchronization URL qualifier, –key.

• MA

MAC using lowercase hexadecimal digits (000e08aabbcc).

• MAU

using uppercase hexadecimal digits (000E08AABBCC).

• MAC

MAC address using lowercase hexadecimal digits with colons to separate pairs of hexadecimal digits (00:0e:08:aa:bb:cc).

• NP

Product name; e.g. SPA2102.

• PSN

; for example 2102.

• SN

Serial number stringfor example 88012BA01234.

• CCERT

SSL, installed or not installed.

• IP

IP address of the SPA phone within its local subnet; e.g. 192.168.1.100.

• EXTIP

external SPA IP, as seen on the Internet; for example 66.43.16.52.

• SWVER

Software version string; for example 2.0.6(b).

• HWVER

Hardware version string; for example 1.88.1.

• PRVST

State provisioning (a numeric string): 

• • -1 = explicit resynchronization request
  • 0 = resynchronization
  • 1 = periodic resynchronization
  • 2 = failed resynchronization, retry

• UPGST

Status update (a numeric string):.

• • 1 = first update attempt
  • 2 = failed update , try again

• UPGERR

Result message (ERR) from the previous update attempt; e.g., http_get failed.

• PRVTMR

Seconds since the last resynchronization attempt. .

• UPGTMR

Seconds since the last update attempt.

• REGTMR1

Seconds since Line 1 lost the registration with the SIP server.

• REGTMR2

Seconds since Line 2 lost registration with the SIP server.

• UPGCOND

Inherited macro name.

• SCHEME

File access scheme (TFTP, HTTP or HTTPS, obtained after parsing the resync or update URL).

• METH

Aliases in disuse for SCHEME, do not use.

• SERV

Requests the host name of the destination server.

• SERVIP

Requests the IP address of the destination server (after DNS lookup). 

• PORT

Request destination UDP/TCP port.

• PATH

Request path of the target file. Request path of the target file.

• ERR

Resynchronization or update attempt result message.

• UIDn.

The contents of the Line user ID configuration parameter n.

• ISCUST

If the unit is customized, value = 1, otherwise 0. .

Note The customization status can be viewed on the information page of the web UI. .

• INCOMINGNAME
• REMOTENUMBER
• DISPLAYNAMEn

The contents of the display name configuration parameter of line N. .

• AUTHIDn

The content of the N-line authentication ID configuration parameter.

For more information on XML support, see the Cisco Small Business Support Community. The URL is provided in Appendix B, “Where to go from here.”

How to configure music on hold on Cisco SPA512G?

To configure the phone to connect to an XML directory service:

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 Enter the following information:

• XML directory service: XML directory name. It is displayed on the user’s phone as a directory option.

• URL of the XML directory service: URL where the XML directory is located.

Step 3 Click Submit all changes.

To configure the phone to connect to an XML application:.

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 Enter the following information:

• service of the XML application: XML application name. It is displayed on the user’s phone as a menu item.

• XML application service: URL where the XML application is located.

If you configured an unused inline button to connect to an XML application, the button connects to the URL configured here, unless you enter a different URL when configuring the inline button. See ” -Horizontal first: same LED flashes with second incoming call.” section on page 2-4.

Step 3 Click Submit all changes.

Your phone can play music on hold if it is part of a system that has a music on hold (MOH) server. To set up music on hold:.

Step 1 Click administrator session > advanced > Voice > Ext_n.

Step 2 In Call Feature Setupin the MOH Server , enter the user ID or URL of the MOH audio streaming server. If you enter a user ID (no server), the current or outbound proxy is contacted.

The default is blank (no MOH). If used with a Cisco SPA9000, the default is imusic. For more information, refer to the Cisco SPA9000 Administration Guide.

Step 3 Click submit all changes.

How to configure SPA512G Extension Mobility?

Extension Mobility allows mobile users to access their phone’s custom settings, such as personal extensions, shared lines and speed dials, from other phones.

For example, people working different shifts or working at different desks during the week can share an extension and still have their own personalized settings.

EM is compatible with BroadSoft and other servers. EM dynamically configures a phone based on the current user. 

A login prompt appears on the IP phone screen when EM is enabled on a phone (e.g., a conference room phone).

A user can enter their user ID and password to access their personal phone settings or ignore the login and use the phone as a guest.

After logging in, users have access to personal directory numbers, services, speed dials and other properties on the phone.

When a user logs out, the phone reverts to a basic profile with limited features enabled. 

To configure extension mobility:.

Step 1 Click admin session > advanced > Voice > Phone.

Step 2 In Extension Mobility, in the Enable EM , choose yes to enable.

Step 3 In the EM User Domain , enter the domain for the phone or authentication server. For example.

@domain.com, which is appended to the user ID ([email protected]) for authentication on the HTTP server.

Step 4 Click Submit all changes. The phone restarts.

You must also configure the Extension Mobility settings in the profile rule field on the Provisioning tab. Refer to the Provisioning Parameters for Extension Mobility on Cisco SPA500 Series IP Phones at:.

https://www.myciscocommunity.com/docs/DOC-11277

For more information on extension mobility and BroadSoft, see http://www.broadsoft .com.

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So much for this administrator’s guide to Configuring Cisco SPA512G. If you want to know how to configure other Cisco terminnals, you can condult this link: Cisco Gesditel Manuals.

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