> For the complete documentation index, see [llms.txt](https://docs.spectralink.net/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://docs.spectralink.net/amie-essentials/applications/biz-phone.md).
# Biz Phone
The Biz Phone app is Spectralink’s SIP telephony application. The following settings may be configured when using a SIP telephony server.
## Enable SIP
Enables/disables the Spectralink SIP application used for call routing, dial plan, DTMF, and URL dialing.
If enabled, the Spectralink SIP application is used for smartphone calls and other telephony services.
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------- |
| **Type** | Radio button |
| **Default** | Disable |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be enabled manually at the Enterprise level for facility-wide telephony functioning |
## SIP Registration 1 & 2
### SIP Registration Server
{% hint style="info" %}
**Formatting servers for failover and redundancy**
Review [Tech Bulletin CS-18-10 *Spectralink Versity Call Server Redundancy*](https://support.spectralink.com/sites/default/files/resource_files/CS-18-10%20Spectralink%20Versity%20Call%20Server%20Redundancy.pdf) when configuring servers for failover and redundancy. Tech Bulletins for Versity 95/96 models can be found at [Spectralink Support](https://support.spectralink.com/s/product-documents/products?type=handsets\&category=wifi).
{% endhint %}
All smartphones using the same SIP server should have the same entry in this field. Two formats are recognized by the smartphone:
**Format A**: domain\
Accepts: IP address, DNS A-name record using a FQDN, or a DNS SRV record that can be resolved to an IP address. The \[domain] value should be resolvable to a server that accepts SIP registrations.
Default value: none
Registration requests, SIP Invites, and other SIP requests will be composed in the form of \[extension]@\[domain] and will be delivered to the IP address of \[domain], whether obtained through DNS lookup or provided by an IP address. If the SIP server requires requests in the form of Extension\@SIP\_Domain\_Name, and you wish to use this format, then you should create a DNS SRV record with the SIP\_Domain\_Name pointing to the server’s address. If you configure a DNS SRV record that returns a port number it will override the port value you configured in the SIP Settings.
**Format B**: \[domain]; \[serverspec1]; \[serverspec2]; \[serverspec3]; \[serverspec4]\
Accepts: domain = SIP Domain Name; serverspec1 = a dotted-decimal IP address, a hostname, or a SRV record that can be resolved to an IP address.
Registration requests, SIP Invites, and SIP messages will be sent in the form of \[extension]@\[domain], will be delivered to the IP address specified after the semicolon, whether obtained through DNS lookup or provided by an IP address. If the SIP server requires requests in the form of \[extension]@\[SIP\_Domain\_Name] and you wish to use this format, you should specify the \[SIP\_Domain\_Name], followed by a semicolon, and then the address, hostname, or SRV record that can be resolved to the address to which you want SIP messages delivered.
**Entry criteria**
| | |
| ------------- | ------------------------------------------------------------------------------------ |
| **Type** | See above for format type |
| **Default** | None |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be set manually at the Enterprise level for facility-wide telephony functioning |
### SIP Registration Server Port
The port value entered here must match the port used in the SIP server to accept connection requests.
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------- |
| **Type** | Integer. Accepts 0, 1 to 65535 |
| **Default** | 5060. If this field is empty, the default is used |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be enabled manually at the Enterprise level for facility-wide telephony functioning |
If utilizing a DNS SRV record, the SIP Server Port should be specified through the SRV record and left blank in this field. If you specify a port number here, then the Versity smartphone will not attempt to use SRV records at all to discover the SIP Server address and instead will only query for a DNS A record. if you configure a DNS SRV record that returns a port number it will override the port value you configured in the SIP Settings.
### SIP Registration Transport
The network transport protocol. Use UDP unless recommended otherwise. (TLS is in development stages. Only server-side TLS has been tested. Contact your Spectralink support agent for further information.)
{% hint style="info" %}
**Which network transport protocol to use?**
Certain call servers drop calls when a phone comes back into Wi-Fi coverage after being out of range. This issue only occurs when using TCP for SIP. *Workaround:* If this is concern in your environment, use UDP for SIP instead of TCP for SIP.
{% endhint %}
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Dropdown menu: UDP, TCP, TLS |
| **Default** | UDP |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### SIP Registration SRTP Enable
May only be enabled if the transport protocol is TLS.
**Entry criteria**
| | |
| ------------- | ------------------------------------------------------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | no |
| **Required?** | |
| **Level** | Enterprise |
| **Notes** | Optional. Only used with TLS. Ignored if another transport method is configured |
### SIP Registration Extension Number
Enter the extension number for the phone. This entry should be different for each device. This field is required and must be populated to achieve successful SIP registration. It is the user part of the SIP Address of Record.
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry. Configurable through .csv |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
Example
If the extension is *1002*, the Address of Record would become <1002@Spectralink.com> where 1002 is the extension number and Spectralink.com is the server value set above.
### SIP Registration Username
SIP digest authentication User ID to be used for SIP registration. For many call servers this will be the same as the Extension Number, though some call servers offer the ability to have a Username that is unique from the Extension Number.
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry. Configurable through .csv |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### SIP Registration Password
The SIP digest authentication password to be used for SIP registration.
Check the **Show Password** check box to see the entry.
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry. Configurable through .csv |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### SIP Expiration Time
The default value is 300.
### SIP Registration Voicemail Retrieval Address
The number to dial to retrieve voicemail messages.
{% hint style="info" %}
**Leave this field blank to allow auto-discovery**
Many call servers will send this address in the Message-Account field of the SIP Notify messages to Spectralink smartphones automatically when the smartphone initially registers.
Leave this field blank to allow the smartphone to attempt auto-discovery of this parameter using the message-account field address provided by the call server.
If long-pressing the number 1 key in the dialer does not initiate a successful call to the voicemail system, populate this field with the number to dial to retrieve a voicemail message (the Voice Mail system’s pilot number.). Invites will be sent to: voicemail\_retrieval\_address\@Server
{% endhint %}
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **Type** | Text entry. ASCII encoded string containing digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL (6416 or <6416@spectralink.com>) |
| **Default** | None |
| **Unique?** | No |
| **Required?** | Maybe, see Admin Tip |
| **Level** | Enterprise |
| **Notes** | Must be set manually at the Enterprise level for facility-wide voicemail functioning
Example 1: if Server is defined as 192.168.20.1 and Voicemail retrieval address is defined as 6416, Invites to voicemail will be sent to 6416\@192.168.20.1
Example 2: If Server is defined as a DNS SRV record that resolves to an IP address (e.g. 192.168.0.1), and Voicemail retrieval address is set to <6416@spectralink.com>, then Invites to voicemail will be sent to the IP address 192.168.0.1 with a To: address of <6416@spectralink.com>
|
### SIP Registration Use Vendor Protocol If Licensed
*Cisco SPP phones only.* Enabled by default, so the phone supports the signaling to work with CUCM. If turned off, all special Cisco signaling is disabled, and the phone works like a non-Cisco SPP phone.
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Enabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | See above |
### SIP Registration Use SIP Standard Hold Signaling
When enabled: Uses SDP media direction parameters (such as a=sendonly) per RFC 3264 when initiating a call.
When disabled: Reverts to RFC2543 hold mechanisms (redirects media to 0.0.0.0 when initiating hold)
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------- |
| **Type** | Radio button |
| **Default** | Enabled |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be enabled manually at the Enterprise level for facility-wide telephony functioning |
{% hint style="success" %}
**When to disable SIP standard hold signaling**
Spectralink recommends leaving this setting enabled unless hold / resume is not functioning correctly on the call server you are attempting to interoperate against.
{% endhint %}
### SIP Registration Force Subscription To Message Waiting Notifications
Many call servers will implicitly subscribe a smartphone to receive message waiting indication (MWI) notifications automatically when they register. Some call servers may require that a smartphone send a separate SIP Subscribe message to receive message waiting notifications. Spectralink recommends that customers leave this setting disabled, unless the Versity smartphone is not receiving message waiting indicators.
Enabled: Causes the smartphone to subscribe to the server address for Message Waiting Notifications
Disabled: The smartphone will not subscribe to the server address for message waiting Notifications
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | See above |
### SIP Registration Allow Contact Header Updates
Allows updates of the call control Contact address using the topmost SIP Via.
Enabled: Will replace the Contact Header with the received IP address and port values contained in the Via of the 200 OK response to SIP Registration. Versity will then renew the SIP Registration using the updated Contact information, and subsequent Invites and Registration requests will be made using the updated Contact information for call control messaging.
Disabled: SIP Message Headers will use the smartphone's local IP address in the Contact field for all call control messaging
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | See above |
### SIP Registration Specify New TCP Port In Contact Header
This parameter is disabled by default.
If enabled, it causes the smartphone to open a new listening port for TCP and put that port number in the contact header field. This parameter should be enabled if the phone will be integrated against a SIP Proxy rather than a SIP (B2BUA) Back to Back User Agent.
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | See above |
### SIP Registration Disable Call Waiting
The administrator can determine whether the phone is allowed to activate call forwarding. If this is turned off, you don't see Call Forwarding in the app menu.
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Enabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | See above |
### Allow Call Forwarding
**Entry criteria**
| | |
| ------------- | ------------------------------- |
| **Type** | Radio button |
| **Default** | Enabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | Not available in Registration 2 |
### Allow Video Calls
*Allow Video Calls* enables the handset to send and receive video calls.
**Entry criteria**
| | |
| ------------- | ------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | Not available in Registration 2 |
### Offer Video in Outgoing Calls
When enabled, a Versity phone will offer to establish a video connection whenever making a call. If the remote end agrees, the video connection is established. When disabled, it will offer to establish a video connection if the user switches to video mode during the call.
**Entry criteria**
| | |
| ------------- | ------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | Not available in Registration 2 |
### Answer Calls in Video Mode
If enabled, a Versity phone will switch to video mode automatically when answering a call offering video. If disabled, the user will have to switch to video mode after the call is answered\*.\*
**Entry criteria**
| | |
| ------------- | ------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise |
| **Notes** | Not available in Registration 2 |
### Send Tones on Local Call Hold
When this setting is enabled a brief tone will sound every five seconds when a call is placed on hold, to confirm to the party on hold that the call is still active. This is useful when working with call servers that do not provide hold tones or hold music.
**Entry criteria**
| | |
| ------------- | ------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | No |
| **Level** | |
| **Notes** | Not available in Registration 2 |
### Server Contact Search
When enabled, Cisco® UCM contacts will be included in contact search results if a phone’s registered call server is a CUCM.
**Entry criteria**
| | |
| ------------- | -------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | Yes, for server contact search |
| **Level** | Enterprise or Group |
| **Notes** | Not available in Registration 2. |
## Common Settings
### Media DSCP
Specify the DSCP of packets. This value should be a decimal number. The default DSCP is 46. Leaving this value blank will enter the default value. This field accepts decimal values from 0 to 63.
**Entry criteria**
| | |
| ------------- | ------------------------------------------------------------------------------------ |
| **Type** | Enter a decimal number Accepts: 0 to 63 (decimal) |
| **Default** | 46 |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be set manually at the Enterprise level for facility-wide telephony functioning |
### Call Control DSCP
Specify the DSCP of packets.
{% hint style="warning" %}
**CAUTION *****Changing Call Control DSCP values requires a smartphone reboot***
Call Control DSCP values do not take effect until the smartphone reboots or SIP service is restarted.
{% endhint %}
**Entry criteria**
| | |
| ------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ |
| **Type** | Enter either decimal (no leading characters) or hex number (0x) Accepts: 0 to 63 decimal or 0x0 to 0x3F hexadecimal (include the 0x if specifying values using hexadecimal notation) |
| **Default** | 40 |
| **Unique?** | No |
| **Required?** | Yes, for telephony functioning |
| **Level** | Enterprise |
| **Notes** | Must be set manually at the Enterprise level for facility-wide telephony functioning |
### Codec Priority
Codecs settings are prioritized by assigning a number to each with 1 being the highest. A setting of 0 (zero) means the codec is disabled. Codecs do not have to be set identically for each smartphone. Options:
* G.711u
* G.711a
* G.722
* G.729a
* Opus
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------- |
| **Type** | Radio button to select and arrows for re-prioritization |
| **Default** | G.711u = 1 and G.711a = 2 The other three codecs are disabled by default |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be enabled manually at the Enterprise level for facility-wide telephony functioning |
### DTMF Relay Payload Type
Spectralink recommends leaving this value at the default setting (96) unless the far end will not accept DTMF payload types of 96 and fails to respond to invites with the service delivery platform (SDP) payload value it requires. Entering a different value allows the RFC2833 DTMF telephone-event payload type advertised in the SDP of SIP messaging to be modified. Value should be a decimal number (no leading characters).
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Dropdown menu. Select 96 to127 |
| **Default** | 96 |
| **Unique?** | No |
| **Required?** | Yes, for telephony functioning |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | |
{% hint style="info" %}
**Telephone-event payload values**
The handset will honor the SDP telephone-event value returned by the called party for DTMF payload negotiation, and will match the SDP telephone-event payload value type of received invites, but modifying this parameter will change the telephone-event payload type the endpoint advertises in initial invites.
{% endhint %}
### Force In-Band DTMF Tones
This parameter is disabled by default. The handset advertises support for RFC2833 through the SIP SDP telephony event parameter.
If enabled, the handset will not advertise support of the telephony event parameters in the SIP SDP, and will result in In-band DTMF tones rather than RFC2833 tones.
{% hint style="info" %}
**Reversion to In-band DTMF**
Versity will revert to In-band DTMF if telephony event packages are not advertised in the SDP of the far end, regardless of the setting of this parameter.
{% endhint %}
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disallow |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Any |
| **Notes** | See above |
### Override Automatic Switch From UDP To TCP
Section 18.1.1 of SIP RFC3261 mandates that SIP packets exceeding 1300 bytes in size should switch their transport protocol from UDP to TCP automatically. If this parameter is enabled, Versity will no longer switch to TCP for packets greater than 1300 bytes. This parameter should be left disabled unless the SIP Server does not honor the RFC or unless you are advised to change this by Spectralink support personnel.
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disallow |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Any |
| **Notes** | See above |
## Cisco Solution Partner Program (SPP)
Cisco-enabled Versity smartphones can use SAM to deploy additional features as programmed in the CUCM server with a COP file. See Appendix A: Spectralink References.
{% hint style="info" %}
**Enable Vendor Protocol option**
Enable the Use Vendor Protocol If Licensed option in the Registration 1 settings to use Cisco configuration options.
{% endhint %}
### Cisco Contact Search
**Entry criteria**
| | |
| ------------- | ----------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | Yes, for contact search |
| **Level** | Enterprise or Group |
| **Notes** | |
### Advanced Contact Search Enable
**Entry criteria**
| | |
| ------------- | -------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | Yes, for advanced contact search |
| **Level** | Enterprise or Group |
| **Notes** | |
### Cisco Server Address
All smartphones using the same SIP server should have the same entry in this field. Two formats are recognized by the smartphone:
**Format A**: domain\
Accepts: IP address, DNS A-name record using a FQDN, or a DNS SRV record that can be resolved to an IP address. The \[domain] value should be resolvable to a server that accepts SIP registrations.
Default value: none
Registration requests, SIP Invites, and other SIP requests will be composed in the form of \[extension]@\[domain] and will be delivered to the IP address of \[domain], whether obtained through DNS lookup or provided by an IP address. If the SIP server requires requests in the form of Extension\@SIP\_Domain\_Name, and you wish to use this format, then you should create a DNS SRV record with the SIP\_Domain\_Name pointing to the server’s address. if you configure a DNS SRV record that returns a port number it will override the port value you configured in the SIP Settings.
**Format B**: \[domain]; \[serverspec1]\
Accepts: domain = SIP Domain Name; serverspec1 = a dotted-decimal IP address, a hostname, or a SRV record that can be resolved to an IP address.
Registration requests, SIP Invites, and SIP messages will be sent in the form of \[extension]@\[domain], will be delivered to the IP address specified after the semicolon, whether obtained through DNS lookup or provided by an IP address. If the SIP server requires requests in the form of \[extension]@\[SIP\_Domain\_Name] and you wish to use this format, you should specify the \[SIP\_Domain\_Name], followed by a semicolon, and then the address, hostname, or SRV record that can be resolved to the address to which you want SIP messages delivered.
**Entry criteria**
| | |
| ------------- | ------------------------------------------------------------------------------------ |
| **Type** | See above for format type |
| **Default** | None |
| **Unique?** | No |
| **Required?** | Yes, for telephony function |
| **Level** | Enterprise |
| **Notes** | Must be set manually at the Enterprise level for facility-wide telephony functioning |
### Cisco Server Username
SIP digest authentication User ID to be used for SIP registration. For many call servers this will be the same as the Extension Number, though some call servers offer the ability to have a Username that is unique from the Extension Number.
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### Cisco Server Password
The SIP digest authentication password to be used for SIP registration.
Check the **Show Password** check box to see the entry.
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### Search Field
**Entry criteria**
| | |
| ------------- | ------------------------------------------------ |
| **Type** | Dropdown menu |
| **Default** | None |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise usually, but may be set at any level. |
| **Notes** | Select search field from one of the options. |
### Search Value
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **Type** | Text entry. |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | No |
| **Level** | Device |
| **Notes** | The string should contain the values of the selected parameter or search field In case of more than one, separate values by a comma(,) *Example:* Hardware, QA, Toys |
### Voicemail Enable
Both traditional and visual voicemail can be deployed with Versity as programmed in the CUCM server with a COP file.
**Entry criteria**
| | |
| ------------- | ------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | Yes for voicemail |
| **Level** | Enterprise or Group |
| **Notes** | |
### Cisco Unity Server Address (Primary)
| | |
| ------------- | ---------------------------------------------------------------------- |
| **Type** | Text field Enter CUCM directory server address starting with https\:// |
| **Default** | None |
| **Unique?** | No |
| **Required?** | Yes |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | |
### Cisco Unity Server Address (Backup)
| | |
| ------------- | ---------------------------------------------------------------------- |
| **Type** | Text field Enter CUCM directory server address starting with https\:// |
| **Default** | None |
| **Unique?** | No |
| **Required?** | Yes |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | |
### Cisco Unity Server Username
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### Cisco Unity Server Password
**Entry criteria**
| | |
| ------------- | -------------------------------------------------------------------- |
| **Type** | Text entry |
| **Default** | None |
| **Unique?** | Yes |
| **Required?** | Yes, for telephony functioning |
| **Level** | Device |
| **Notes** | Must be set at the Device level for smartphone telephony functioning |
### Allow iDivert
This feature allows the user to redirect (or divert) calls to the Voicemail system as programmed in the CUCM server with a COP file.
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------------------------------------------- |
| **Type** | Radio button |
| **Default** | Enabled |
| **Unique?** | No |
| **Required?** | Yes for voicemail |
| **Level** | Enterprise |
| **Notes** | See the *Versity Applications Administration Guide* and *Spectralink Versity: Cisco SPP SIP Integration: CUCM with COP file* |
### Allow Call Park
Cisco supports several variants of the Call Park feature as programmed in the CUCM server with a COP file.
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------------------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | See |
| **Level** | Enterprise |
| **Notes** | See the *Versity Applications Administration Guide* and *Spectralink Versity: Cisco SPP SIP Integration: CUCM with COP file* |
### Allow Hunt Group Login/Logout
Hunt groups can be configured to allow the user to log in and log out. Enable this option if you want hunt group members to have this choice.
**Entry criteria**
| | |
| ------------- | ---------------------------------------------------------------------------------------------------------------------------- |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise or Group |
| **Notes** | See the *Versity Applications Administration Guide* and *Spectralink Versity: Cisco SPP SIP Integration: CUCM with COP file* |
## Call Park Settings
Call parking lets a user put a call on hold and then retrieve the call from another phone. When a call is parked it is transferred to a call park extension. The parked call can then be retrieved by calling that extension.
Call parking is disabled by default. If enabled (by setting *Allow Call Park* to **True**), an administrator can configure up to 5 different call park extensions.
The extension number is entered into the *Call Park Extension* field.
Some call servers require a prefix to be appended to an extension number in order to retrieve a parked call. In that circumstance the prefix and extension number can be entered into the optional *Call Retrieval Extension* field. (If this field is left blank the Call Park Extension number will be used to retrieve the parked call.)
See the *Versity Applications Administration Guide* for more information on configuring and using call park.
### Allow Call Park
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise |
| **Notes** | |
### Call Park Extension 1-5 Enable
**Entry criteria**
| | |
| ------------- | ------------ |
| **Type** | Radio button |
| **Default** | Disabled |
| **Unique?** | |
| **Required?** | No |
| **Level** | Enterprise |
| **Notes** | |
### Call Park Extension 1-5
**Entry criteria**
| | |
| ------------- | ---------- |
| **Type** | Text entry |
| **Default** | None |
| **Unique?** | |
| **Required?** | No |
| **Level** | Enterprise |
| **Notes** | |
### Call Retrieval Extension 1-5
This optional field should be left blank unless your call server requires a prefix to be appended to an extension number in order to retrieve a parked call.
**Entry criteria**
| | |
| ------------- | --------------------- |
| **Type** | Text entry |
| **Default** | None |
| **Unique?** | |
| **Required?** | No |
| **Level** | Enterprise |
| **Notes** | See description above |
## Emergency Location
These new settings support RedSky HELD (HTTP-Enabled Location Delivery) integration. HELD assigns physical locations to APs, so that a caller’s location can be determined based on the AP covering their call. These settings only apply if your system is integrated with RedSky.
{% hint style="danger" %}
**Emergency Location services**
The reliability of Emergency Location services depends on the functionality and reliability of your greater infrastructure which includes factors beyond Spectralink's control, including but not limited to:
* Network infrastructure (wireless LAN, LAN, call servers)
* Central provisioning and configuration management servers
* Integration of third-party location services (including RedSky integration)
* Central security systems and servers
* Correct configuration of Spectralink Versity smartphones
* Correct installation and system integration
* Personnel training and operational procedures
Emergency Location services are not fault-tolerant and must not be your sole solution for emergency response. You must maintain alternative emergency procedures and safe practices.
Use of Emergency Location services is subject to all terms, disclaimers, and limitations of liability set forth in the Spectralink End User License Agreement (EULA), including but not limited to the High Risk Activities provisions. Spectralink assumes no responsibility and shall not be liable for any failure, interruption, or inaccuracy of Emergency Location services.
{% endhint %}
### Emergency HELD Company ID
Enter the ID assigned by RedSky.
**Entry criteria**
| | |
| ------------- | --------------------- |
| **Type** | Text field |
| **Default** | None |
| **Unique?** | |
| **Required?** | |
| **Level** | Enterprise |
| **Notes** | See description above |
### Emergency HELD URL
Enter the URL of the PBX configured for HELD.
**Entry criteria**
| | |
| ------------- | --------------------- |
| **Type** | Text field |
| **Default** | None |
| **Unique?** | |
| **Required?** | |
| **Level** | Enterprise |
| **Notes** | See description above |
### Emergency HELD Location Enable
Must be set to *True* if using Emergency HELD Location.
**Entry criteria**
| | |
| ------------- | --------------------- |
| **Type** | Radio button |
| **Default** | False |
| **Unique?** | |
| **Required?** | |
| **Level** | Enterprise |
| **Notes** | See description above |
## User Settings
The administrator can limit or allow access to User Settings through the UI State field which can make an option Disabled or Not Visible. See the *Versity Applications Administration Guide* for more information.
## LDAP
{% hint style="warning" %}
**LDAP settings are highly specialized!**
Prior to configuring the settings listed here, consult your LDAP administrator to obtain correct information for populating these settings.
{% endhint %}
You can connect your smartphone to a corporate directory server that supports the Lightweight Directory Access Protocol (LDAP) version 3. Once configured, the corporate directory can be used by the smartphone.
Versity smartphones support the following LDAP servers:
* Microsoft® Active Directory®
* Open LDAP Directory Server 2.4.12
Spectralink smartphones support corporate directories that support server-side sorting and those that do not. For servers that do not support server-side sorting, sorting is performed on the smartphone.
{% hint style="info" %}
**Better performance with server-side sorting**
Spectralink recommends using corporate directories that have server-side sorting for better performance. Consult your LDAP Administrator when making any configuration changes for the corporate directory. For more information on LDAP attributes, see [RFC 4510 - Lightweight Directory Access Protocol (LDAP): Technical Specification Road Map](http://tools.ietf.org/html/rfc4510).
{% endhint %}
{% hint style="info" %}
**Supported LDAP directories**
Configuration of a corporate directory depends on the LDAP server you use. For detailed explanations and examples of all currently supported LDAP directories, see Technical Bulletin 41137: *Best Practices When Using Corporate Directory on Spectralink handsets*.
{% endhint %}
### *Server Address*
The IP address or hostname of the LDAP server interface to the corporate directory. For example, *host.domain.com*.
**Entry criteria**
| | |
| ------------- | ----------------------------------------------------------------- |
| **Type** | Text field Accepts: dotted-decimal IP address or hostname or FQDN |
| **Default** | None |
| **Unique?** | No |
| **Required?** | Yes |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Server Port*
The port that connects to the server if a full URL is not provided. Common LDAP ports are 389 and 636 for encrypted (TLS).
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text field Accepts 0, Null, 1 to 65535 |
| **Default** | 389 |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Communication Security Type*
If you are using encryption, select the type: SSL, None, Start TLS
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Dropdown menu |
| **Default** | None |
| **Unique?** | No |
| **Required?** | See above |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Bind DN*
Distinguished Name for the site. An LDAP representation of the name of your domain. You may include the CN for the admin.
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text field |
| **Default** | None \[blank] (leave empty for entire site) |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Bind PW*
For OpenLDAP, a password is used to access the directory.
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text field |
| **Default** | None \[blank] (leave empty for anonymous) |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Base DN*
Use the same value as the Bind DN. However, if you have a space or other special characters in the Bind DN name use encoded characters here.
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text field |
| **Default** | None \[blank] (leave empty for entire site) |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Primary Email Attribute*
In your implementation of LDAP, this is the attribute you are using for the primary email addresses of your contacts (for example, *mail* or *msExchxxx*).
{% hint style="warning" %}
**The “mail” attribute should be validated against an email server**
The “mail” attribute may be a text field on the LDAP server and might not contain a valid email address. Be sure to verify email addresses against an email server to ensure that valid email addresses are being provided to smartphones.
{% endhint %}
{% hint style="warning" %}
**Use attributes that are populated by a single email address only**
Certain LDAP attributes may contain multiple emails, such as a primary address and a number of alias addresses. Use an attribute that provides only a single email address.
{% endhint %}
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text field |
| **Default** | none |
| **Unique?** | No |
| **Required?** | Yes if you are using LDAP |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
### *Alternate Email Attribute*
A different field used for email addresses. For instance, you might have multiple mail servers, or some contacts may use some other email address.
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text field |
| **Default** | none |
| **Unique?** | No |
| **Required?** | Yes if you are using LDAP |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
## Emergency Contacts
{% hint style="danger" %}
**Power failure may prevent call functionality**
A Versity smartphone may not function in the event of a power failure as it relies on the computer network to transmit calls. You will have a registration failure icon if the smartphone cannot make or receive calls, but there may be a time delay between the system failure and the appearance of the registration failure icon.
{% endhint %}
When a smartphone is locked, it cannot be used for emergency calls unless emergency contact numbers are configured here. This is a safety feature that many facilities deploy to protect their employees. Up to 5 numbers can be configured.
### Contact Name, Contact Number
Up to five emergency contacts can be configured with both a name and a number.
Ensure you follow any dial plan rules when entering the number.
**Entry criteria**
| | |
| ------------- | ----------------------------------------------- |
| **Type** | Text fields |
| **Default** | None |
| **Unique?** | No |
| **Required?** | No |
| **Level** | Enterprise usually, but may be set at any level |
| **Notes** | See above |
---
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