Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide

Cadence Script

A CadScript is a mini-script that specifies the cadence parameters of a signal up to 127 characters in length. Syntax:

where:

Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/offi,5[,oni,6/offi,6]]]]]) and is known as a section

oni,j and offi,j are the on/off durations in seconds of a segment. The subscript ranges are: i = 1 or 2, j = 1 to 6.

Di is the total duration of the section in seconds. All durations can have up to three decimal places to provide 1 ms resolution. The wildcard character “*” indicates infinite duration.

The segments within a section are played in order and repeated until the total duration is played.

Example: Normal Ring

where:

• Number of Cadence Sections = 1
• Cadence Section 1: Section Length = 60 s
• Number of Segments = 1
• Segment 1: On=2s, Off=4s
• Total Ring Length = 60s

Example 2: Distinctive Ring (short, short, short, long)

where:

• Number of Cadence Sections = 1
• Cadence Section 1: Section Length = 60s
• Number of Segments = 4
• Segment 1: On=0.2s, Off=0.2s
• Segment 2: On=0.2s, Off=0.2s
• Segment 3: On=0.2s, Off=0.2s
• Segment 4: On=1.0s, Off=4.0s
• Total Ring Length=60s

Tone Script

A ToneScript is a mini-script that specifies the frequency, level, and cadence of a call progress tone. It can contain up to 127 characters. Syntax:

Section Zi is similar to the Si section in a CadScript except that each on/off segment is followed by a frequency parameter: Zi = Di(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2 [,oni,3/offi,3/fi,3 [,oni,4/offi,4/fi,4 [,oni,5/offi,5/fi,5 [,oni,6/offi,6/fi,6]]]]])

where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] 1< nk < 6 indicates which of the frequency components given in the FreqScript are used in that segment; if more than one frequency component is used in a segment, the components are summed together.

Example: Dial Tone

• Number of Frequencies = 2
• Frequency 1 = 350 Hz at –19 dBm
• Frequency 2 = 440 Hz at –19 dBm
• Number of Cadence Sections = 1
• Cadence Section 1: Section Length = 10 s
• Number of Segments = 1
• Segment 1: On=forever, with Frequencies 1 and 2
• Total Tone Length = 10s
• Example 2: Stutter Tone
• 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)
• Number of Frequencies = 2
• Frequency 1 = 350 Hz at –19 dBm
• Frequency 2 = 440 Hz at –19 dBm
• Number of Cadence Sections = 2
• Cadence Section 1: Section Length = 2s
• Number of Segments = 1
• Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2
• Cadence Section 2: Section Length = 10s
• Number of Segments = 1
• Segment 1: On=forever, with Frequencies 1 and 2
• Total Tone Length = 12s

Example: SIT Tone

• Number of Frequencies = 3
• Frequency 1 = 985 Hz at –16 dBm
• Frequency 2 = 1428 Hz at –16 dBm
• Frequency 3 = 1777 Hz at –16 dBm
• Number of Cadence Sections = 1
• Cadence Section 1: Section Length = 20s
• Number of Segments = 4
• Segment 1: On=0.38s, Off=0s, with Frequency 1
• Segment 2: On=0.38s, Off=0s, with Frequency 2
• Segment 3: On=0.38s, Off=0s, with Frequency 3
• Segment 4: On=0s, Off=4s, with no frequency components
• Total Tone Length = 20s

Ring Script

A RingScript is a mini-script that describes a ring tone. Syntax:

ring-tone-name identifies this ring tone. This name appears on the Ring Tone menu of the phone. The same name can be used in a SIP Alert-Info header in an inbound INVITE request to tell the phone to play the corresponding ring tone. The name should contain the same characters allowed in a URL only.

waveform-id-or-path is the index of the desired waveform to use in this ring tone. The built-in waveforms are:

– 1 = Classic phone with mechanical bell

– 2 = Typical phone ring

– 3 = Classic ring tone

– 4 = Wide-band frequency sweep signal

This field can also be a network path (url) to download a ring tone data file from a server on-the-fly. In this case, the syntax of the field is:

cadence-id is the index of the desired cadence to play the given waveform. 8 cadences (1–8) as defined in <Cadence 1> through <Cadence 8>. Cadence-id can be 0 If w=3,4, or an url. Setting c=0 implies the on-time is the natural length of the ring tone file.

break-time specifies the number of seconds to break between two bursts of ring tone, such as b=2.5.

total-time specifies the total number of seconds to play the ring tone before it times out.

Example 1: Normal Ring

where:

• Number of Cadence Sections = 1
• Cadence Section 1: Section Length = 60 s
• Number of Segments = 1
• Segment 1: On=2s, Off=4s
• Total Ring Length = 60s

Example 2: Distinctive Ring (short, short, short, long)

where:

• Number of Cadence Sections = 1
• Cadence Section 1: Section Length = 60s
• Number of Segments = 4
• Segment 1: On=0.2s, Off=0.2s
• Segment 2: On=0.2s, Off=0.2s
• Segment 3: On=0.2s, Off=0.2s
• Segment 4: On=1.0s, Off=4.0s
• Total Ring Length=60s

Entering Star Code Values

The phone provides default values for star codes. To change star code values, navigate to Admin Login > advanced > Regional. Under Vertical Service Activation Codes, enter the values you want to change for the codes.

The codes are:

• Call Return (*69)—Calls the last caller, regardless which extension.
• Blind Transfer (*98)—Allows the user to transfer a call to another number without waiting for the other party to pick up.
• Call Back Act (*66)—Periodically redials the last busy number (every 30 seconds by default) until it rings or until the attempt expires (30 min by default), regardless which extension. Only one call back operation can be ordered at a time. A new order automatically cancels the previous order.
• Call Back Deact (*86)—Cancels the last call back operation.
• Call Forward All Act (*72)—Call forwards all inbound calls. Applies to primary extension only.
• Call Forward All Deact (*73)—Cancels call forward all. Applies to primary extension only.0
• Call Forward Busy Act (*90)—Call forwards on busy. Applies to primary extension only.
• Call Forward Busy Deact (*91)—Cancels call forward on busy. Applies to primary extension only.
• Call Forward No Answer Act (*92)—Call forwards if no answer. Applies to primary extension only.
• Call Forward No Answer Deact (*93)—Cancels call forward no answer. Applies to primary extension only.
• CW Act (*56)—Enables call waiting. For example, if call waiting is turned off globally, this star code will turn on call waiting until the CW Deact code is entered.
• CW Deact (*57)—Deactivates call waiting. For example, if call waiting is turned on globally, this star code deactivates call waiting until the CW Act code is entered.
• CW Per Call Act (*71)—Enables call waiting for a single call. For example, if call waiting is turned off globally, this star code will turn on call waiting for that call.
• CW Per Call Deact (*70)—Deactivates call waiting for a single call. For example, if call waiting is turned on globally, this star code deactivates call waiting for that call.
• Block CID Act (*67)—Blocks caller ID on all outbound calls. Applies to all extensions.
• Block CID Deact (*68)—Deactivates caller ID blocking on outbound calls. Applies to all extensions.
• Block CID Per Call (*81)—Blocks caller ID on the next outbound call (on the current call appearance only).
• Block CID Per Call Deact (*82)—Deactivates caller ID blocking on the next outbound call (on the current call appearance only).
• Block ANC Act—Blocks anonymous calls. Applies to all extensions.
• Block ANC Deact—Deactivates anonymous call blocking. Applies to all extensions.
• DND Act (*78)—Activates Do Not Disturb. Applies to all extensions.
• DND Deact (*79)—Deactivates Do Not Disturb. Applies to all extensions.
• Secure All Call Act (*16)—Defaults to prefer to use encrypted media (voice codecs).
• Secure No Call Act (*17)—Defaults to prefer to use unencrypted media for all outbound calls. Applies to all extensions.
• Secure One Call Act (*18)—Prefers to use encrypted media for the outbound call (on this call appearance only).
• Secure One Call Deact (*19)—Prefers to use unencrypted media for the outbound call (on this call appearance only).
• Paging (*96)—Pages the number called.
• Call Park (*38)—Parks a call on an entered line number.
• Call UnPark Code (*39)—Retrieves a call from an entered line number.
• Call Pickup (*36)—Picks up a call at an entered extension.
• Group Call Pickup (*37)—Picks up a ringing call at a group of extensions.
• Media Loopback Code (*03)—A service provider can set up a test call from an IP media loopback server (the source) to a subscriber VoIP device (the mirror). The test call provides statistical reporting on network performance and audio quality.
  
  Depending on the source capabilities, a service provider can see packet jitter, loss, and delay (although Media Loopback cannot identify an offending hop). This helps the service provider identify an offending hop that could be causing issues in VoIP calls to a subscriber. The test results can also provide audio quality scoring, that lets a service provider better understand the subscriber’s experience.
• Referral Services Codes—One or more * codes can be configured into this parameter, such as *98, or *97|*98|*123, and so forth. The maximum total length is 79 characters.
  
  This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone. Each * code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the Cisco IP phone to perform a blind transfer to a target number that is prepended by the service * code. For example:

– After the user dials *98, the Cisco IP phone plays a special prompt tone while waiting for the user the enter a target number (which is validated according to the dial plan as in normal dialing).

– When a complete number is entered, the Cisco IP phone sends a blind REFER to the holding party with the Refer-To target equals to *98 target_number. This feature allows the Cisco IP phone to hand off a call to an application server to perform further processing, such as call park.

The * codes should not conflict with any of the other vertical service codes internally processed by the Cisco IP phone. You can delete any * code that you do not want the call server to process.

Feature Dial Services Codes: Tells the Cisco IP phone what to do when the user is listening to the first or second dial tone.

You can configure one or more * codes into this parameter, such as *72, or *72|*74|*67|*82, and so on. The maximum total length is 79 characters. When the user has a dial tone (first or second dial tone), they can enter a * code (and the following target number according to current dial plan) to trigger the Cisco IP phone to call the target number prepended by the * code. For example:

– After the user dials *72, the Cisco IP phone plays a special prompt tone while waiting for the user the enter a target number (which is validated according to the dial plan as in normal dialing).

– When a complete number is entered, the Cisco IP phone sends an INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features such as call forward (*72) or BLock Caller ID (*67).

You can add a parameter to each * code in Features Dial Services Codes to indicate what tone to play after the * code is entered, such as *72‘c‘|*67‘p‘.
Following is a list of allowed dial tone parameters (note the use of back quotes surrounding the parameter without spaces).

– ‘c‘ = Cfwd dial tone

– ‘d‘ = Dial tone

– ‘m‘ = MWI dial tone

– ‘o‘ = Outside dial tone

– ‘p‘ = Prompt dial tone

– ‘s‘ = Second dial tone

– ‘x‘ = No tones are place, x is any digit not used above

If no tone parameter is specified, the Cisco IP phone plays the prompt tone by default.

If the * code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include it in this parameter. In that case, add that * code in the dial plan.

Activating or Deactivating Supplementary Services

You can disable services handled locally by the phone in one of two ways:

• Delete the star code in the Vertical Service Activation section in the Regional tab.
• Disable the service in the Phone tab. See Configuring Supplementary Services (Star Codes).

If a service is enabled in the Phone tab but cleared in the Regional tab, the service can still be enabled/disabled from the IP phone screen or the phone web user interface. If a service is disabled, the soft button associated with that service is hidden on the IP phone screen. Also, any menu item associated with a disabled service is preceded with an exclamation mark (!).

A supplementary service should be disabled if

• the user has not subscribed to it
• the service provider intends to support similar service by using means other than relying on the Cisco IP phone.

Bonus Services Announcement Description

When the user enables the callback service using the *code, the IP phone automatically calls 123400@$proxy.

When the user disables the callback service using the *code, the IP phone automatically connects to the 123401@$proxy.

If the Service Annc Extension Codes do not define CBT/CBF extension codes, the IP phone does not use this feature.

Outbound Call Codec Selection Codes

Codec call selection codes affect voice quality. For more information about voice codecs, see the Configuring Voice Codecs. You can choose a preferred codec for a call or force a call to use a specific codec:

• Prefer G.711u (*017110) through G.729a (*01729)—Sets the preferred codec for next outbound call. If the preferred codec is unavailable, the second, then the third preferred codec is used, if specified.
• Force G.711u (*027110) through G.729a (*02729) —Forces the specified codec for next outbound call. If the specified codec is unavailable, the preferred codecs are used in order, if specified.

See Configuring Voice Codecs for more information.

DTMF Parameters

DTMF is used by touch-tone phones to assign a specific frequency (consisting of two separate tones) to each key so that it can easily be identified by a microprocessor.

In-Band and Out-of-Band (RFC-2833): IP phones can relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information.

The following parameters can either reduce false detection or get better detection by the IVR. In general, the default values are recommended for both IVR functions.

• DTMF Playback Level: Local DTMF playback level in decibels per minute, up to one decimal place. Applicable locally when a user presses a digit or when the phone receives an out-of-band (OOB) DTMF signal from the network side. Does not affect DTMF transmission. Defaults to -16.
• DTMF Playback Length: Local DTMF playback duration in milliseconds. Affects only OOB. Defaults to.1.
• Inband DTMF Boost: Controls the amount of amplification applied to DTMF signals. Affects only tones sent by inband method. Choices are 0, 3, 6, 9, 12, 15, and 18 decibels. Defaults to 12 dB.

To support false detection, avoid inband and use OOB. With OOB, the DTMF Playback Length does not matter. If you use inband, use a smaller DTMF Boost value.

To get better detection by the IVR, avoid inband and use OOB. This way, the DTMF tone is reconstructed by the PSTN gateway or the remote endpoint, and the quality is not subject to distortion from the audio codec. If you use OOB, use a slightly longer DTMF Playback Length.

If you use inband, use a higher Inband DTMF boost.

Note On the Cisco SPA525G2, when using the Mobile Link line (through the Bluetooth-enabled mobile phone), the local user can hear a double tone (echo) when pressing digits (DTMF tones) and engaged on a call. This can happen with certain mobile phones that have the option to play locally the local tone (which is also played by the Cisco SPA525G2). This does not affect operation with interactive voice response applications, as the tone is audible only on the local device. See the Cisco support community at http://www.cisco.com/go/smallbizsupport for phone compatibility information, and also consult the latest Cisco SPA525G2 release notes, available at cisco.com.

Managing the Time and Date

Cisco IP phones obtain the time settings in one of three ways:

• NTP Server—When the phone boots up, it tries to contact the first Network Time Protocol (NTP) server to get the time. The phone periodically synchronizes its time with the NTP server. The synchronization period is fixed at 1 hour. Between updates the phone tracks time with its internal clock.
• SIP Messages—Each SIP message (request or response) sent to the phone could contain a Date header with the current time information. If the header is present, the phone uses it to set its clock.
• Manual Setup—The time and date can be entered manually by using the IP phone screen or the phone web user interface. However, this value is overwritten by the NTP time or SIP Message Date whenever they are available to the phone. Manual setup requires that you enter the time in 24-hour format only.

The time served by the NTP Server and the SIP Date Header are expressed in GMT time. The local time is obtained by offsetting the GMT according to the time zone of the region.

The Time Zone parameter can be configured by using the phone web user interface or through provisioning. This time can be further offset by the Time Offset (HH/mm) parameter. This parameter must be entered in 24-hour format and can also be configured from the IP phone screen.

The Time Zone and Time Offset (HH/mm) offset values are not applied to manual time and date setup.

Configuring Daylight Saving Time

The phone supports auto adjustment for daylight saving time. You must set Daylight Savings Time Enable to yes and enter the DST rule. This option affects the time stamp on the CallerID.

To enter the rule for calculating DST, include the start, end, and save values separated by semi-colons (;) as follows:

For example, the default DST rule is:

The start-time and end-time values specify the start and end dates and times of daylight saving time. The format is:

The month value equals any value in the range 1-12 (January-December).

The day value equals any + or - value in the range 1-31. If value is -1, the time will change on the weekday on or before the end of the month; the last occurrence of a weekday in that month.

The weekday value equals any value in the range -7 to 7 (Monday to Sunday). If the weekday value is 0, the date to start or end daylight saving is exactly the month and day. If the weekday value is -7 to 7, daylight saving starts or ends on the weekday value on or after the month and day. If the weekday value is not 0 and the day value is negative, then daylight saving starts or ends on the weekday value on or before the month and day.

Optional time values: HH represents hours (0-23), mm represents minutes (0-59). and ss represents seconds (0-59). Optional values inside brackets [ ] are assumed to be 0 if not specified. Midnight is represented by 0:0:0.

The save-time value is the number of hours, minutes, and/or seconds to add to the current time during DST. The save-time value cannot be a minus (-) in version 7.5.1 and higher. Earlier firmware versions support a negative value.

Daylight Saving Time Examples

The following example configures daylight saving time for the U.S, adding one hour starting at midnight on the first Sunday in April and ending at midnight on the last Sunday of October; add 1 hour (USA, North America):

The following example configures daylight saving time for Egypt, starting at midnight on the last Sunday in April and ending at midnight on the last Sunday of September:

The following example configures daylight saving time for New Zealand (in version 7.5.1 and higher), starting at midnight on the first Sunday of October and ending at midnight on the third Sunday of March.

The following example reflects the new change starting March 2007. DST starts on the second Sunday in March and ends on the first Sunday in November:

Selecting a Display Language

This section describes how to localize the Cisco SPA300 Series and Cisco SPA500 Series IP Phone display language. You can define up to nine languages, in addition to English, to be available and host the dictionaries for each of the languages on the HTTP or TFTP provisioning server. Language support follows Cisco dictionary principles.

Note The Cisco WIP310 does not support localization.

Use the Language Selection parameter to select the phone default display language. The value must match one of the languages supported by the dictionary server. The script (dx value) is as follows:

• <Language_Selection ua="na">
• </Language_Selection>

Defaults to blank; the maximum number of characters is 512. For example:

During startup, the phone checks the selected language and downloads the dictionary from the TFTP/HTTP provisioning server indicated in the phone configuration. The dictionaries are available at the support website. See Appendix B, “Where to Go From Here,” for the website location.

The end user can change the language of the phone on the phone by following these steps:

Step 1 Press the Setup button.

Step 2 Select Language, then press the Select soft button.

Step 3 Select Option to change the language.

Step 4 With the desired language highlighted, press Save.

Creating a Dictionary Server Script

The Dictionary Server Script defines the location of the dictionary server, the languages available and the associated dictionary. The syntax is:

Note TFTP, HTTP, and HTTPS are supported for the dictionary download.

Defaults to blank; the maximum number of characters is 512. The detailed format is as follows:

For example: