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TopicSeniorIpPhone (SDT3)
Description
This resource allows to connect a Topic Senior IP Phones.
Release notes
Version 1.2.0 - Early deployment
- Bugfix: DHCP Option 120 should be disabled by default when provisioned (TS-5087)
- Dependency:
- Baseline 2.x.x
- Topic Support >= 1.1.0
- Feature: Ability to enable and configure LLDP interval (M14133)
- Deprecated: DHCP Option 120 should be disabled by default when provisioned (TS-5087)
- Dependency:
- Baseline 2.x.x
- Topic Support >= 1.0.0
- Feature: initial version (M11802)
- Deprecated: DHCP Option 120 should be disabled by default when provisioned (TS-5087)
- Dependency:
- Baseline 2.x.x
- Topic Support >= 1.0.0
Resource configuration interface
GUI unavailable.
Resource configuration parameters
This resource allows to configure the Topic Senior IP phone D312ID.- Inter-digit timeout: Number of seconds to wait before placing the call without the need to press the (re)dial button. This value is set to 5 seconds by default.
- Speed Dial, from left to right:
- Softkey 1: Extension to call when pressing the 1st softkey.
- Softkey 2: Extension to call when pressing the 2nd softkey.
- Softkey 3: Extension to call when pressing the 3rd softkey.
- SIP server feature: Note: These technical settings if improperly configured can have impact on the availability of the phone
- SIP re-invite: allow the voice media to be transferred directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
- Default: Keep default settings as defined in the Telephony server module (for example Asterisk 1.2.x)
- Yes: Allow the server to perform the re-invite procedure.
- No: Do not allow the server to perform the re-invite procedure.
- Qualify SIP devices registration: In order to monitor the SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
- Default: Keep the default setting defined in the Telephony server module (for example Asterisk 1.2.x)
- No: Disable Qualify SIP registration for this device
- Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
- Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
- Force specific codec: If this parameter is set to 'Yes', the SIP server will only accept the 3 codecs defined in this resource with the specified preference.
- First Codec, Second Codec, Third Codec: Ordered list of codec
- SIP re-invite: allow the voice media to be transferred directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
Limitations
The following are not supported by this phone.- Corporate directory or call history
- Active active support
- Call transfer
- PUM, COLP, ZeroConf, Video
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