faderline_out_gainmeter_peak_hold_modemeter_peak_resetmic_in_gainmutephantomsafety_mutesignal_activity_threshtrimphone_auto_answer_enphone_connectphone_dialphone_dial_backspacephone_dial_tone_gainphone_dtmf_gainphone_entry_tone_enphone_exit_tone_enphone_flashphone_flash_delayphone_ignorephone_redialphone_rejectphone_ringphone_ring_tonephone_ring_tone_enphone_tone_gainpstn_auto_hangup_loop_enpstn_auto_hangup_call_prog_enpstn_countrypstn_dtmf_tone_durationpstn_flash_delay_overridepstn_in_gainpstn_line_voltagepstn_loop_currentpstn_out_gainvoip_answervoip_blindvoip_board_infovoip_bootblock_sw_vervoip_bootrom_sw_vervoip_call_appearancevoip_call_appearance_infovoip_call_appearance_linevoip_call_appearance_statevoip_cancelvoip_conferencevoip_dhcp_boot_servvoip_dhcp_boot_serv_optvoip_dhcp_boot_serv_typevoip_dhcp_option_60_typevoip_dial_modevoip_dndvoip_eth_settingsvoip_eth_vlan_idvoip_factory_resetvoip_forwardvoip_holdvoip_joinvoip_linevoip_line_labelvoip_line_statevoip_loc_cityvoip_loc_company_namevoip_loc_countryvoip_loc_descriptionvoip_loc_house_numbervoip_loc_house_number_suffixvoip_loc_locationvoip_loc_postal_codevoip_loc_post_directionalvoip_loc_pre_directionalvoip_loc_statevoip_loc_street_namevoip_loc_street_suffixvoip_local_resetvoip_message_waitingvoip_net_cfg_savevoip_notificationvoip_popupvoip_prov_serv_addressvoip_prov_serv_passwordvoip_prov_serv_typevoip_prov_serv_uservoip_rebootvoip_resumevoip_sendvoip_splitvoip_statusvoip_transfervoip_uc_sw_vereq_eneq_typegeq_compensategeq_engeq_gaingeq_typehigh_shelf_enhigh_shelf_frequencyhigh_shelf_gainhigh_shelf_slopehorn_enhorn_frequencyhpf_enhpf_frequencyhpf_orderhpf_typelow_shelf_enlow_shelf_frequencylow_shelf_gainlow_shelf_slopelpf_enlpf_frequencylpf_orderlpf_typepeq_band_enpeq_bandwidthpeq_frequencypeq_gainpeq_typedp_endp_gate_attackdp_gate_decaydp_gate_endp_gate_holddp_gate_ratiodp_gate_threshdp_exp_attackdp_exp_decaydp_exp_endp_exp_ratiodp_exp_threshdp_comp_attackdp_comp_decaydp_comp_endp_comp_ratiodp_comp_threshdp_lim_attackdp_lim_decaydp_lim_endp_lim_ratiodp_lim_threshdp_peak_endp_peak_threshaec_enaec_noise_fillaec_refagc_enagc_max_gainagc_min_gaindelaydelay_enfb_enfb_filter_bandwidthfb_filter_decay_enfb_filter_resetfb_filter_max_depthfb_safe_mode_attenmic_source_indexmic_source_typenc_ennc_levelsig_gen_gainsig_gen_sweep_startsig_gen_sweep_stepsig_gen_sweep_stopsig_gen_sweep_timesig_gen_tone_freqsig_gen_typesmartpairing_ensmartpairing_dev_idam_adapt_thresham_camera_activity_timeam_chairmanam_chan_biasam_decay_timeam_enam_gain_sharingam_groupam_hold_timeam_last_mic_modeam_nom_limitam_off_attenam_priorityam_priority_attenam_slopeanalog_gpio_valuedigital_gpio_helddigital_gpio_hold_timedigital_gpio_repeat_timedigital_gpio_statedigital_gpio_valueauth_passwordclink_call_activeclink_local_call_activeclink_muteclink_num_video_channelsclink_volumeeth_auth_modeeth_settingseth_macir_key_pressir_key_heldir_chan_idser_baudser_control_modeser_flowser_sendclink_num_attacheddev_bootloader_verdev_firmware_verdev_hw_ecodev_hw_revdev_led_cycledev_ntp_serverdev_plugin_typedev_statusdev_tempdev_temp_statusdev_tp_modedev_typedev_uptimedev_volt_clinkdev_volt_neg_15dev_volt_phantomdev_volt_pos_15sys_bus_idsys_cmd_log_ensys_devices_matchsys_factory_resetsys_last_full_presetsys_last_partial_presetsys_last_presetsys_meter_periodsys_mtrreg_log_ensys_namesys_num_auth_connectionssys_num_connectionssys_num_devssys_num_open_connectionssys_pausesys_plugins_matchsys_rebootsys_tp_cfgsys_last_tp_cfgThe fader, gain, and mute parameters are described here. The telephony gains, faders, and mutes are described in the Telephony Parameters section. The matrix crosspoint gains and mutes are described in the Matrix Parameters section.
fader Digital Fader
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | Yes |
| Event Source | Yes |
This parameter sets the fader level (in dB) in the digital domain.
| Command | Response | Description |
|---|---|---|
| set fader "Amplifier" 1 |
val fader "Amplifier" 1 |
Sets the fader on the "Amplifier" virtual channel to 1. |
| get fader "Amplifier" |
val fader "Amplifier" 1 |
Returns the current value of the fader on the "Amplifier" virtual channel. |
| set fader max "Amplifier" 10 |
val fader max "Amplifier" 10 |
Sets the maximum fader value to +10 on the "Amplifier" virtual channel. Any commands to set the fader above +10 will have the value set to 10. |
| set fader min "Amplifier" -15 |
val fader min "Amplifier" -15 |
Sets the minimum fader value to -15 on the "Amplifier" virtual channel. Any commands to set the fader below -15 will have the value set to -15. |
| inc fader "Amplifier" 2 |
val fader "Amplifier" 3 |
Increments the current value of the fader by 2dB on the "Amplifier" virtual channel and returns the current value of the fader. |
| inc fader "Amplifier" 0.5 |
val fader "Amplifier" 3.5 |
Increments the current value of the fader by 0.5dB on the "Amplifier" virtual channel and returns the current value of the fader. |
| dec fader "Amplifier" 2.5 |
val fader "Amplifier" 1 |
Decrements the current value of the fader by 2.5dB on the "Amplifier" virtual channel and returns the current value of the fader. |
line_out_gain Line Output Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 4.0, Resolution: 0.5 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) of the line output.
| Command | Response | Description |
|---|---|---|
| set line_out_gain "Amplifier" -10 |
val line_out_gain "Amplifier" -10 |
Sets the line_out_gain on the output channel "Amplifier" to -10dB. |
meter_peak_hold_mode Meter Peak Hold Mode
| Channel Type | Global System | |||||||||
| Value Type | List | |||||||||
| Read/Write Mode | Read/Write | |||||||||
| Values |
|
|||||||||
| Event Source | No |
This parameter defines the peak hold behavior of all of the peak
meters in the system. If the infinite peak hold mode is selected, the
peaks can be reset using the meter_peak_reset
parameter.
meter_peak_reset Meter Peak Reset
| Channel Type | Global System |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Event Source | No |
This parameter resets all of the peak meters in the system, if the peak meters are configured to have the infinite peak hold behavior.
mic_in_gain Mic Input Pre-Amp Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input, Telepresence Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 64.0, Resolution: 0.5 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) of the mic pre-amp. A separate mic/line control is not provided. Instead, a continuous gain range is provided, and the firmware will map this to the appropriate mic/line switch and pre-amp gain settings.
| Command | Response | Description |
|---|---|---|
| set mic_in_gain "Table Mic 1" 48 |
val mic_in_gain "Table Mic 1" 48 |
Sets the analog preamp gain to 48dB for input channel "Table Mic 1". |
mute Digital Mute
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Signal Generator, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | Yes |
This parameter sets the mute status of the virtual
channel. A value of 0 indicates the virtual
channel is unmuted, while a value of 1 indicates
it is muted.
| Command | Response | Description |
|---|---|---|
| set mute "Amplifier" 1 |
val mute "Amplifier" 1 |
Sets the mute on the "Amplifier" virtual channel to true -- the channel is muted. |
| set mute "Amplifier" 0 |
val mute "Amplifier" 0 |
Sets the mute on the "Amplifier" virtual channel to false -- the channel is unmuted (i.e., not muted). |
| get mute "Table Mic 1" |
val mute "Table Mic 1" 1 |
Queries the mute status on the "Table Mic 1" virtual channel to see if the channel is muted. The value 1 means it is muted, 0 means it is not muted |
| set mute "Mics" 1 |
val mute "Table Mic 1" 1 val mute "Table Mic 2" 1 val mute "Table Mic 3" 1 val mute "Table Mic 4" 1 val mute "Mics" 1 |
Query the current value of the mute of the virtual channel group "Mics" which has four microphones as group members, "Table Mic 1", "Table Mic 2", "Table Mic 3", and "Table Mic 4". |
phantom 48V Phantom Power
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input, Telepresence Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
Enables or disables phantom power on mic
inputs. Setting phantom to 1
enables phantom power, while setting it to 0
disables phantom power.
| Command | Response | Description |
|---|---|---|
| set phantom "Table Mic 1" 1 |
val phantom "Table Mic 1" 1 |
Enables the 48V phantom power supply for the input channel "Table Mic 1". |
| set phantom "Table Mic 1" 0 |
val phantom "Table Mic 1" 0 |
Disables the 48V phantom power supply for the input channel "Table Mic 1". |
safety_mute Safety Mute
| Channel Type | Global System |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Default | 0 |
| Event Source | Yes |
This parameter sets the status of the safety mute. If
safety mute is enabled (1), all line outputs of
all the devices are muted.
| Command | Response | Description |
|---|---|---|
| set safety_mute 1 |
val safety_mute 1 |
Enables the safety_mute for a SoundStructure system. |
| set safety_mute 0 |
val safety_mute 0 |
Turns off the safety_mute for a SoundStructure system. |
signal_activity_thresh Signal Activity Threshold
| Channel Type | Global System |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| System Limits | Minimum: -100.0, Maximum: 20.0, Resolution: 0.1 |
| Default | -20.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the threshold in dBU for the signal activity meter. The signal activity meter is based on the VU signal meter.
| Command | Response | Description |
|---|---|---|
| get signal_activity_thresh |
get signal_activity_thresh -20 |
This returns the current value for the signal activity threshold. |
| set signal_activity_thresh 0 |
val signal_activity_thresh 0 |
Sets the signal_activity_thresh to 0. If the VU meter value of the signal crosses 0dB then the signal will be reported as active. |
trim Gain Trim for Virtual Channels
| Channel Type | Virtual Channel | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | |||
| Virt Chans | Stereo | |||
| Indices |
|
|||
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.5 | |||
| Default | 0.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter applies gain (in the analog domain) to the
individual components of a virtual channel. The index
indicates to which physical channel of the virtual channel the
trim will be applied. For example, indices 1 and
2 correspond to the left and right physical
channels of a stereo virtual channel.
| Command | Response | Description |
|---|---|---|
| set trim "Program Audio" 1 2 |
val trim "Program Audio" 1 2 |
Sets the trim value of the left channel (channel 1) of the stereo virtual channel "Program Audio" to 2dB. |
| set trim "Program Audio" 2 -3 |
val trim "Program Audio" 2 -3 |
Sets the trim value of the right channel (channel 2) of the stereo virtual channel "Program Audio" to -3dB. |
matrix_balance Matrix Crosspoint Balance
| Channel Type | Matrix |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Submix, ConferenceLink Aux Input, ConferenceLink Raw Input, Telepresence Mic/Line Input, Telepresence Line Input, Multichannel to Stereo Downmixer Input |
| Row Virt Chans | Stereo |
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, Multichannel to Stereo Downmixer Output, ConferenceLink Raw Output |
| Col Virt Chans | Mono, Stereo |
| System Limits | Minimum: -1.0, Maximum: 1.0, Resolution: 0.01 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
The matrix_balance parameter is available at
crosspoints where stereo virtual channels are mixed to mono or
stereo virtual channels. The matrix_balance
parameter provides a way to control the amount of gain going
to the left and right channels.
| Command | Response | Description |
|---|---|---|
| set matrix_balance "Program Audio" "Codec Line Mix Out" 1 |
val matrix_balance "Program Audio" "Codec Line Mix Out" 1.000 |
Sends only the right channel of the stereo program audio "Program Audio" source to both stereo audio outputs "Codec Line Mix Out" |
| set matrix_balance "Program Audio" "Amplifier" 0 |
val matrix_balance "Program Audio" "Amplifier" 0.000 |
Sets the balance so that left is sent to left and right is sent to right on the stereo virtual channel input and output. This ensures the stereo program audio "Program Audio" left and right channels are sent to the stereo audio output "Amplifier" left and right channels, respectively. |
matrix_gain Matrix Crosspoint Gain
| Channel Type | Matrix |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Signal Generator, Submix, ConferenceLink Aux Input, ConferenceLink Raw Input, Telepresence Mic/Line Input, Telepresence Line Input, Stereo to Multichannel Upmixer Input, Multichannel to Stereo Downmixer Input |
| Row Virt Chans | Mono, Stereo |
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, Multichannel to Stereo Downmixer Output, ConferenceLink Raw Output |
| Col Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | Yes |
| Event Source | No |
This parameter sets the gain (in dB) for the specified crosspoint in the matrix mixer.
| Command | Response | Description |
|---|---|---|
| set matrix_gain "Table Mic 1" "Phone Out" 3 |
val matrix_gain "Table Mic 1" "Phone Out" 3 |
Sets the matrix crosspoint gain from input "Table Mic 1" to output "Phone Out" to 3dB. |
| get matrix_gain "Table Mic 1" "Phone Out" |
val matrix_gain "Table Mic 1" "Phone Out" 3 |
Queries the matrix crosspoint gain from input "Table Mic 1" to output "Phone Out". |
matrix_gate Enable Gated Signal at Crosspoint
| Channel Type | Matrix |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Row Virt Chans | Mono, Stereo |
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, Multichannel to Stereo Downmixer Output, ConferenceLink Raw Output |
| Col Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter selects whether the gated (1)
or ungated (0) version of the input signal is
sent to the output.
| Command | Response | Description |
|---|---|---|
| set matrix_gate "Table Mic 1" "Phone Out" 1 |
val matrix_gate "Table Mic 1" "Phone Out" 1 |
Tells the matrix to use the automixed version of the input signal "Table Mic 1" when creating the output signal "Phone Out". |
| set matrix_gate "Table Mic 1" "Phone Out" 0 |
val matrix_gate "Table Mic 1" "Phone Out" 0 |
Tells the matrix to use the un-automixed version of input "Table Mic 1" when creating the output signal "Phone Out". |
matrix_gate_type Select Gating Type
| Channel Type | Matrix | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input | ||||||
| Row Virt Chans | Mono, Stereo | ||||||
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, Multichannel to Stereo Downmixer Output, ConferenceLink Raw Output | ||||||
| Col Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter selects the gating style for crosspoints
with microphone inputs. Gating is enabled with the
matrix_gate parameter.
| Command | Response | Description |
|---|---|---|
| set matrix_gate_type "Table Mic 1" "Phone Out" conf |
val matrix_gate_type "Table Mic 1" "Phone Out" conf |
Tells the matrix to use the conferencing version of the input processing of the input signal "Table Mic 1" when creating the output signal "Phone Out". |
| set matrix_gate_type "Table Mic 1" "Amplifier" sr |
val matrix_gate_type "Table Mic 1" "Amplifier" sr |
Tells the matrix to use the sound reinforcement version of the input processing of the input signal "Table Mic 1" when creating the output signal "Amplifier". |
matrix_invert Matrix Crosspoint Inversion
| Channel Type | Matrix |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Signal Generator, Submix, ConferenceLink Aux Input, ConferenceLink Raw Input, Telepresence Mic/Line Input, Telepresence Line Input, Stereo to Multichannel Upmixer Input, Multichannel to Stereo Downmixer Input |
| Row Virt Chans | Mono, Stereo |
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, Multichannel to Stereo Downmixer Output, ConferenceLink Raw Output |
| Col Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
Inverts the specified crosspoint in the matrix
mixer. Setting matrix_invert to 0
sets the crosspoint to normal polarity; setting
matrix_invert to 1 inverts the
crosspoint polarity.
| Command | Response | Description |
|---|---|---|
| set matrix_invert "Table Mic 1" "Phone Out" 1 |
val matrix_invert "Table Mic 1" "Phone Out" 1 |
Inverts (negates) the crosspoint from "Table Mic 1" to the output "Phone Out". "Table Mic 1" will still be heard by the remote participants on "Phone Out" but now has been inverted (negated). |
| get matrix_invert "Table Mic 1" "Phone Out" |
val matrix_invert "Table Mic 1" "Phone Out" 1 |
Queries the invert status of the crosspoint from "Table Mic 1" to the output channel apos;Phone Out". A value of 1 means the crosspoint has been inverted. A value of zero means the crosspoint value has not been inverted. |
matrix_mute Matrix Crosspoint Mute
| Channel Type | Matrix |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Signal Generator, Submix, ConferenceLink Aux Input, ConferenceLink Raw Input, Telepresence Mic/Line Input, Telepresence Line Input, Stereo to Multichannel Upmixer Input, Multichannel to Stereo Downmixer Input |
| Row Virt Chans | Mono, Stereo |
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, Multichannel to Stereo Downmixer Output, ConferenceLink Raw Output |
| Col Virt Chans | Mono, Stereo |
| Default | 1 |
| Event Source | No |
Mutes or unmutes the specified crosspoint in the matrix
mixer. Setting matrix_mute to 0
unmutes the crosspoint; setting matrix_mute to
1 mutes the crosspoint.
| Command | Response | Description |
|---|---|---|
| set matrix_mute "Table Mic 1" "Phone Out" 1 |
val matrix_mute "Table Mic 1" "Phone Out" 1 |
Mutes the crosspoint from input "Table Mic 1" to the output "Phone Out" so "Table Mic 1" will not be heard by the remote participants on "Phone Out". |
| get matrix_mute "Table Mic 1" "Phone Out" |
val matrix_mute "Table Mic 1" "Phone Out" 1 |
Queries the mute status of the crosspoint from "Table Mic 1" to the output channel apos;Phone Out". |
matrix_pan Matrix Crosspoint Pan
| Channel Type | Matrix |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Row Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Signal Generator, Submix, ConferenceLink Aux Input, ConferenceLink Raw Input, Telepresence Mic/Line Input, Telepresence Line Input, Stereo to Multichannel Upmixer Input, Multichannel to Stereo Downmixer Input |
| Row Virt Chans | Mono |
| Col Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Submix, ConferenceLink Aux Output, Telepresence Line Output, Stereo to Multichannel Upmixer Output, ConferenceLink Raw Output |
| Col Virt Chans | Stereo |
| System Limits | Minimum: -1.0, Maximum: 1.0, Resolution: 0.01 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
The matrix_pan parameter is available at
crosspoints where mono virtual channels are mixed to stereo
virtual channels. The matrix_pan parameter
provides a way to control the amount of gain going to the left
and right channels.
| Command | Response | Description |
|---|---|---|
| set matrix_pan "Phone In" "Amplifier" 1 |
val matrix_pan "Phone In" "Amplifier" 1.000 |
Sends the mono virtual channel channel "Phone In" to only the right channel of the stereo virtual channel "Amplifier". |
| set matrix_pan "Phone In" "Amplifier" -1 |
val matrix_pan "Phone In" "Amplifier" -1.000 |
Sends the mono virtual channel "Phone In" to only the left channel of the stereo virtual channel "Amplifier". |
| set matrix_pan "Phone In" "Amplifier" 0 |
val matrix_pan "Phone In" "Amplifier" 0.000 |
Sends the mono virtual channel "Phone In" to both the left and right channels of the stereo virtual channel "Amplifier". |
| set matrix_pan "Table Mic 1" "Codec Stereo Mics Out" -1 |
val matrix_pan "Table Mic 1" "Codec Stereo Mics Out" -1.000 |
Sends the mono virtual channel "Table Mic 1" to the left channel of the stereo virtual channel "Codec Stereo Mics Out". |
phone_auto_answer_en Enable Auto-Answer for Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input, VoIP Telephony Input |
| Virt Chans | Mono |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the auto-answer feature for the telephony
interface.
| Command | Response | Description |
|---|---|---|
| set phone_auto_answer_en "Phone In" 1 |
val phone_auto_answer_en "Phone In" 1 |
Sets the phone associated with the virtual channel "Phone In" to autoanswer when the phone rings. Note that the phone in virtual channel name must be used, not the phone output virtual channel name. |
phone_connect Connect or Disconnect Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output, VoIP Telephony Output |
| Virt Chans | Mono |
| Default | 0 |
| Event Source | Yes |
This parameter sets the connection status of the
telephony interface. Setting the phone_connect
status to 1 connects the call, while setting it
to 0 disconnects the call.
| Command | Response | Description |
|---|---|---|
| set phone_connect "Phone Out" 1 |
val phone_connect "Phone Out" 1 |
Takes the phone output channel "Phone Out" offhook. Note that the phone out virtual channel name must be used, not the phone input virtual channel name. |
| get phone_connect "Phone Out" 0 |
val phone_connect "Phone Out" 0 |
Hangs up the phone line associated with the virtual channel "Phone Out". Note that the phone out virtual channel name must be used, not the phone input virtual channel name. |
phone_dial Dial the Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Write-Only |
| Phys Chans | Analog Telephony Output, VoIP Telephony Output |
| Virt Chans | Mono |
| System Limits | Max String Length: 128 |
| Event Source | No |
This command dials the specified string of characters on the telephony interface. This parameter can be used to dial one digit at a time or many digits all at once. For the PSTN interface, valid digits are '0' through '9', '*', '#', and ',' (delay). For the VoIP interface, all characters are valid.
If the telephony interface is not connected (see the
phone_connect parameter) when this parameter
is set, then the characters will be stored in a dial
buffer. When the telephony interface is eventually connected,
the characters stored in the dial buffer shall be dialed. If
more than 20 seconds pass after the last
phone_dial or phone_connect
parameters are sent, then the dial buffer is automatically
cleared.
| Command | Response | Description |
|---|---|---|
| set phone_dial "Phone Out" "9,18009322774" |
val phone_dial "Phone Out" "9,18009322774" |
Dials the phone line associated with the virtual channel "Phone Out" with the digit string "9,18009322774". The phone line must be offhook for the digits to be dialed - see phone_connect. Note that the phone out virtual channel name must be used, not the phone input virtual channel name. |
phone_dial_backspace Delete Previously Dialed Digit
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | Analog Telephony Output, VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter deletes the most recently added
character from the phone_dial dial buffer. If
there are no characters in the dial buffer, then setting this
parameter has no effect. For the PSTN interface, this
parameter only affects the dial buffer when the phone
interface is on-hook. This is true for the VoIP interface as
well, but the parameter also affects the dial buffer when the
interface is off-hook before a call is placed.
phone_dial_tone_gain Dial tone gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input |
| Virt Chans | Mono |
| System Limits | Minimum: -100.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter controls the gain that is applied to the incoming phone signal when dial tone is present.
| Command | Response | Description |
|---|---|---|
| set phone_dial_tone_gain "Phone In" -6 |
val phone_dial_tone_gain "Phone In" -6 |
Sets the gain of the dial tone heard in the room for the phone associated with the virtual channel "Phone In" to -6dB when the phone is taken offhook. Note that the phone in virtual channel name must be used, not the phone output virtual channel name. |
phone_dtmf_gain Telephony Input DTMF Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) applied to DTMF
tones generated to the local room. To adjust the level of
ring tones, entry tones, and exit tones played back into the
local room, use the phone_tone_gain
parameter.
| Command | Response | Description |
|---|---|---|
| set phone_dtmf_gain "Phone In" -6 |
val phone_dtmf_gain "Phone In" -6 |
Sets the level of the dtmf digits that are heard in the local room from the phone interface associated with the virtual channel "Phone In" to -6dB. Note that the phone in virtual channel name must be used, not the phone output virtual channel name. |
phone_entry_tone_en Enable Entry Tones for Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input, VoIP Telephony Input |
| Virt Chans | Mono |
| Default | 1 |
| Event Source | No |
This parameter enables or disables entry tone generation
for the telephony interface. If entry tones are enabled
(1), then an entry tone is played whenever the
auto-answer feature engages and connects the telephony
interface. Entry tones and exit tones (see the
phone_exit_tone_en parameter) are typically
enabled to prevent a caller from entering or exiting a
conference unannounced.
| Command | Response | Description |
|---|---|---|
| set phone_entry_tone_en "Phone In" 0 |
val phone_entry_tone_en "Phone In" 0 |
Turns off the entry tone for virtual channel "Phone In". This means if phone_auto_answer_en is set to 1 there will be no sound generated in the local room when the phone is auto answered. Note that the phone in virtual channel name must be used, not the phone output virtual channel name. |
phone_exit_tone_en Enable Exit Tones for Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input, VoIP Telephony Input |
| Virt Chans | Mono |
| Default | 1 |
| Event Source | No |
This parameter enables or disables exit tone generation
for the telephony interface. If exit tones are enabled
(1), then an exit tone is played whenever the
auto-hangup feature engages and disconnects the telephony
interface. Entry tones (see the
phone_entry_tone_en parameter) and exit tones
are typically enabled to prevent a caller from entering or
exiting a conference unannounced.
| Command | Response | Description |
|---|---|---|
| set phone_exit_tone_en "Phone In" 0 |
val phone_exit_tone_en "Phone In" 0 |
Turns off the exit tone for virtual channel "Phone In". This means if if the phone auto hangs up there will be no sound generated in the local room when the phone is hung up. Note that the phone in virtual channel name must be used, not the phone output virtual channel name. |
phone_flash Connect or Disconnect Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter disconnects the telephony
interface then reconnects it after a short delay. The amount
of delay can be configured with the
phone_flash_delay parameter.
| Command | Response | Description |
|---|---|---|
| set phone_flash "Phone Out" |
val phone_flash "Phone Out" |
Flashes the phone interface associated with the virtual channel "Phone Out". Note that the phone out virtual channel name must be used, not the phone in virtual channel name. |
phone_flash_delay Set Flash Delay
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: 10, Maximum: 5000 |
| Default | 100 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the delay (in milliseconds) for the
phone_flash parameter. Note that by default,
PSTN interfaces use the flash delay determined by their
pstn_country setting. However, they can use
the value of this parameter if the
pstn_flash_delay_override parameter is set to
1.
phone_ignore Ignore an Incoming Call
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | Analog Telephony Output, VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter causes the incoming call to be ignored. The incoming ring tones will be silenced on the local side, but the remote caller will still hear ringing.
| Command | Response | Description |
|---|---|---|
| set phone_ignore "Phone Out" |
val phone_ignore "Phone Out" |
Tells the SoundStructure to ignore the phone and stop the ringing. The remote caller still hears the phone ring, but it is silenced in the local room. |
phone_redial Redial the Last Number on the Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | Analog Telephony Output, VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter causes the last number to be
redialed on the telephony interface. The last number is
defined as all the digits that were dialed since the telephony
interface was last connected (see the
phone_connect parameter). If the telephony
interface is not already connected, setting this parameter
will automatically connect it before dialing.
| Command | Response | Description |
|---|---|---|
| set phone_redial "Phone Out" |
val phone_redial "Phone Out" |
Dials the last number dialed on the virtual channel "Phone Out". |
phone_reject Reject an Incoming Call
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | Analog Telephony Output, VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter causes the incoming call to be rejected. For the VoIP interface, the call is rejected and immediately sent to voicemail. For the PSTN interface, the call is terminated by automatically answering it, then immediately hanging up (the audio paths remain muted so that the conference is not interrupted and so that no conference audio goes to the incoming call).
phone_ring Ring Indicator for Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read-Only |
| Phys Chans | Analog Telephony Input, VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | Yes |
This parameter indicates the ringing state for the
telephony interface. While the telephony interface is
ringing, reading this parameter will return 1.
When the telephony interface is not ringing, reading this
parameter will return 0.
Acknowledgements for this parameter will be automatically sent whenever this parameter changes state due to a hook flash, auto-answer, or auto-hangup.
| Command | Response | Description |
|---|---|---|
| val phone_ring "Phone In" 1 |
Returns the value 1 when the incoming phone line associated with the virtual channel "Phone In" is ringing. Note that the phone in virtual channel name must be used, not the phone out virtual channel name. |
phone_ring_tone Select Ring Tone for Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Sequence |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input, VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Minimum: 1, Maximum: 14 |
| Default | 1 |
| Event Source | No |
This parameter selects the type of ring tone to be generated for the telephony interface. If ring tones are enabled, the selected tone is played whenever there is an incoming ring signal on the telephony interface.
The parameter values correspond to the following tones.
| Command | Response | Description |
|---|---|---|
| set phone_ring_tone "Phone In" 3 |
val phone_ring_tone "Phone In" 3 |
Sets the phone_ring_tone for the "Phone In" channel to 3, low double trill. Please note this parameter operates on the "Phone In" channel. |
phone_ring_tone_en Enable Ring Tones for Telephony Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input |
| Virt Chans | Mono |
| Default | 1 |
| Event Source | No |
This parameter enables or disables ring tone generation
for the telephony interface. If ring tones are enabled
(1), then a ring tone is played whenever there is
an incoming ring signal on the telephony interface.
| Command | Response | Description |
|---|---|---|
| set phone_ring_tone_en "Phone In" 0 |
val phone_ring_tone_en "Phone In" 0 |
Turns off the ring tone that is normally generated when an incoming call is received on virtual channel "Phone In". This can be used with control system applications to have a silent ring in the room for privacy. Note that the phone in virtual channel name must be used, not the phone output virtual channel name. |
phone_tone_gain Telephony Input Tone Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input, VoIP Telephony Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) applied to tones
generated to the local room. In particular, this gain applies
to the ring tone, entry tone, and exit tone. To adjust the
level of the DTMF digits played back to the local room, use
the phone_dtmf_gain parameter.
For the VoIP interface, this parameter only controls the entry and exit tone gains; it does not control the ring tone gain.
pstn_auto_hangup_loop_en Enable Auto-Hangup on Loop Drop for PSTN Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the auto-hangup on loop drop feature for the
PSTN interface.
| Command | Response | Description |
|---|---|---|
| set pstn_auto_hangup_loop_en "Phone Out" 1 |
val pstn_auto_hangup_loop_en "Phone Out" 1 |
Sets the auto hangup based on loop current to be enabled on the phone channel "Phone Out". |
pstn_auto_hangup_call_prog_en Enable Auto-Hangup on Call Progress for PSTN Interface
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the auto-hangup on call progress tones
feature for the PSTN interface.
| Command | Response | Description |
|---|---|---|
| set pstn_auto_hangup_call_prog_en "Phone Out" 1 |
val pstn_auto_hangup_call_prog_en "Phone Out" 1 |
Sets the auto hangup based on call progress (busy, fast busy, offhook) to be enabled on the phone channel "Phone Out". Please note this parameter operates on the "Phone Out" channel. |
pstn_country Country for PSTN Interface
| Channel Type | Virtual Channel | |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Value Type | List | |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Read/Write Mode | Read/Write | |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Phys Chans | Analog Telephony Output | |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Virt Chans | Mono | |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Values |
|
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| Event Source | No |
This parameter configures the PSTN interface for operation in a specific country.
| Command | Response | Description |
|---|---|---|
| set pstn_country "Phone Out" canada |
val pstn_country "Phone Out" canada |
Sets the country code for the telephony interface "Phone Out" to canada. |
pstn_dtmf_tone_duration Tone duration for DTMF tones
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: 10, Maximum: 600000 |
| Default | 100 |
| User Limits Supported | No |
| Event Source | No |
This parameter controls the duration (in milliseconds) of the tone generated for each DTMF digit.
| Command | Response | Description |
|---|---|---|
| set pstn_dtmf_tone_duration "Phone Out" 300 |
val pstn_dtmf_tone_duration "Phone Out" 300 |
Sets the duration of tones generating each DTMF digit to 300msec for the analog telephony interface associated with the channel "Phone Out". |
pstn_flash_delay_override Override country code flash delay
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| Default | 0 |
| Event Source | No |
This parameter controls whether or not the flash hook delay
is determined by the default pstn_country
settings (0) or by the phone_flash_delay
setting (1).
pstn_in_gain PSTN Input Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Input |
| Virt Chans | Mono |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.5 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) of the signal coming from the PSTN interface.
| Command | Response | Description |
|---|---|---|
| set pstn_in_gain "Phone In" 6 |
val pstn_in_gain "Phone In" 6 |
Adjusts the input gain on the phone input to 6dB. Note that the phone in virtual channel name must be used, not the phone out virtual channel name. |
pstn_line_voltage PSTN Line Voltage
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: -128, Maximum: 128 |
| Event Source | No |
This parameter indicates the line voltage (in Volts) of the PSTN interface. The value is valid in both on-hook and off-hook modes. The value can be positive or negative, indicating the polarity of the tip/ring voltage. When the value changes sign, it indicates that a polarity reversal has occurred.
| Command | Response | Description |
|---|---|---|
| get pstn_line_voltage "Phone Out" |
val pstn_line_voltage "Phone Out" 0 |
Queries the pstn_line_voltage and returns the measured value in Volts on the phone line "Phone Out". Note that the phone out virtual channel name must be used, not the phone in virtual channel name. |
pstn_loop_current PSTN Loop Current
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read-Only |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: 0.0, Maximum: 281.6, Resolution: 0.1 |
| Event Source | No |
This parameter indicates the loop current (in milliamps) of the PSTN interface. The value is only valid when the interface is off-hook.
pstn_out_gain PSTN Output Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.5 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) of the signal going to the PSTN interface.
voip_answer Answer an Incoming Call
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used to answer an incoming call while the VoIP interface is currently in a different call.
| Command | Response | Description |
|---|---|---|
| set voip_answer "VoIP Out" |
val voip_answer "VoIP Out" |
This command answers an incoming call on the "VoIP" channel. If there is no incoming call, this command does not affect the system. If you are already on a call, this command will place the existing call on hold and answer an incoming call. |
voip_blind Specify a Blind Transfer
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used along with voip_transfer to make a blind transfer.
| Command | Response | Description |
|---|---|---|
| set voip_blind "VoIP Out" |
val voip_blind "VoIP Out" |
This command is sent after voip_transfer command to turn the transfer into a blind transfer where the active call appearance on the "VoIP" channel is transferred to another extension without waiting for the second extension to be answered. |
voip_board_info Get Board Info
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Event Source | No |
This parameter returns manufacturing and hardware information about the VoIP plug-in card.
voip_bootblock_sw_ver Get Bootblock Software Version
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Event Source | No |
This parameter returns the bootblock version of the VoIP plug-in card.
voip_bootrom_sw_ver Get Bootrom Software Version
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Event Source | No |
This parameter returns the bootrom version of the VoIP plug-in card.
voip_call_appearance Select the Active Call Appearance
| Channel Type | Virtual Channel |
| Value Type | Sequence |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: 1, Maximum: 24 |
| Default | 1 |
| Event Source | No |
This parameter selects the currently active call
appearance. Some parameters (e.g. phone_connect, phone_dial, phone_redial) operate on the currently active call
appearance, as specified by this parameter. Setting this
parameter is analogous to selecting a call appearance on the
UI of a Polycom VoIP phone.
| Command | Response | Description |
|---|---|---|
| set voip_call_appearance "VoIP Out" 2 |
val voip_call_appearance "VoIP Out" 2 |
Selects the second call appearance on the channel "VoIP Out". |
voip_call_appearance_info Call Appearance Info
| Channel Type | Virtual Channel | ||||||
| Value Type | String | ||||||
| Read/Write Mode | Read-Only | ||||||
| Phys Chans | VoIP Telephony Output | ||||||
| Virt Chans | Mono | ||||||
| Indices |
|
||||||
| System Limits | Max String Length: 128 | ||||||
| Event Source | No |
This parameter reports textual information for the specified call appearance. There are two lines of textual information that can be independently queried via the second index to this parameter. Typically, the two lines of information are the local and remote caller ID or number.
voip_call_appearance_line Call Appearance Line Number
| Channel Type | Virtual Channel | |||
| Value Type | Integer | |||
| Read/Write Mode | Read-Only | |||
| Phys Chans | VoIP Telephony Output | |||
| Virt Chans | Mono | |||
| Indices |
|
|||
| System Limits | Minimum: 1, Maximum: 12 | |||
| Event Source | No |
This parameter reports the line number associated with the specified call appearance.
voip_call_appearance_state Call Appearance State
| Channel Type | Virtual Channel | ||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Value Type | List | ||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Read/Write Mode | Read-Only | ||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Phys Chans | VoIP Telephony Output | ||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Virt Chans | Mono | ||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Indices |
|
||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Values |
|
||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Event Source | No |
This parameter reports the call appearance state for the specified call appearance. Automatic status messages are generated for this parameter when it changes automatically.
| Command | Response | Description |
|---|---|---|
| get voip_call_appearance_state "VoIP Out" 2 |
val voip_call_appearance "VoIP Out" connected |
Gets the state of the call appearance on the channel "VoIP Out". The state value will depend on the state of the call. Note that this command is read-only. |
voip_cancel Cancel a Transfer or Conference
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used to cancel a transfer or conference.
| Command | Response | Description |
|---|---|---|
| set voip_cancel "VoIP Out" |
val voip_cancel "VoIP Out" |
Tells the system to cancel the transfer or conference that was initiated for channel "VoIP Out" with either a voip_transfer or a voip_conference command. |
voip_conference Start a Conference Call
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used to start a conference call.
| Command | Response | Description |
|---|---|---|
| set voip_conference "VoIP Out" |
val voip_conference "VoIP Out" |
When in a call, voip_conference wll put the current call associated with channel "VoIP Out" on hold and generate dialtone to support dialing a second caller. Once the second call is established, a second set voip_conference "VoIP Out" is required to merge the cfalls together. See the example in the SoundStructure Design Guide for Dialing two calls on the same line. |
voip_dhcp_boot_serv Set boot server option for VoIP card
| Channel Type | Virtual Channel | ||||||||||||
| Value Type | List | ||||||||||||
| Read/Write Mode | Read/Write | ||||||||||||
| Phys Chans | VoIP Telephony Input | ||||||||||||
| Virt Chans | Mono | ||||||||||||
| Values |
|
||||||||||||
| Event Source | No |
This parameter controls the boot server option for the VoIP interface.
voip_dhcp_boot_serv_opt Set boot server option number for VoIP card
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Minimum: 0, Maximum: 255 |
| Default | 150 |
| User Limits Supported | No |
| Event Source | No |
When voip_dhcp_boot_serv is set to
custom, this parameter specifies the DHCP option
number in which the VoIP card will look for the boot server.
| Command | Response | Description |
|---|---|---|
| set voip_dhcp_boot_serv_opt "VoIP In" 168 |
val voip_dhcp_boot_serv_opt "VoIP In" 168 |
Sets the DHCP option number to specify the field that the VoIP interface will use for the boot server information for the channel "VoIP In". Note that this command operates on the VoIP Input channel. |
voip_dhcp_boot_serv_type Set boot server option type for VoIP card
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | VoIP Telephony Input | ||||||
| Virt Chans | Mono | ||||||
| Values |
|
||||||
| Event Source | No |
When voip_dhcp_boot_serv is set to
custom, this parameter specifies the type of the
DHCP option in which the VoIP card will look for the boot
server.
| Command | Response | Description |
|---|---|---|
| set voip_dhcp_boot_serv_type "VoIP In" ip_address |
val voip_dhcp_boot_serv_type "VoIP In" ip_address |
Sets the boot server type of the DHCP option to ip_address for the channel "VoIP In". Note that this command operates on the VoIP Input channel. |
| set voip_dhcp_boot_serv_type "VoIP In" string |
val voip_dhcp_boot_serv_type "VoIP In" string |
Sets the boot server type of the DHCP option to string for the channel "VoIP In". |
voip_dhcp_option_60_type Set boot server option 60 type for VoIP card
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | VoIP Telephony Input | ||||||
| Virt Chans | Mono | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter specifies the format for the vendor identifying information used with a DHCP server when DHCP option 60 is enabled.
| Command | Response | Description |
|---|---|---|
| set voip_dhcp_option_60_type "VoIP In" rfc3925_binary |
val voip_dhcp_option_60_type "VoIP In" rfc3925_binary |
Specifies the format of the vendor identifying information to rfc3925_binary for the channel "VoIP In". |
| set voip_dhcp_option_60_type "VoIP In" ascii_string |
val voip_dhcp_option_60_type "VoIP In" ascii_string |
Sets the boot server type of the DHCP option to ascii_string for the channel "VoIP In". |
voip_dial_mode Set dial mode for VoIP card
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | VoIP Telephony Output | ||||||
| Virt Chans | Mono | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter is used to select between number dialing and
SIP URL dialing. Whenever the phone_connect
parameter for a voip_out channel changes from
1 to 0, the voip_dial_mode parameter shall be reset back to
number. An automatic status message shall be
generated when this happens.
| Command | Response | Description |
|---|---|---|
| set voip_dial_mode "VoIP Out" number |
val voip_dial_mode "VoIP Out" number |
Sets the dial mode to number. Digits dialed will be assumed to be numbers from a standard dialpad. |
| set voip_dial_mode "VoIP Out" url |
val voip_dial_mode "VoIP Out" url |
Sets the dial mode to url. This allows the user to dial IP addresses instead of digits. |
voip_dnd Enable or disable do-not-disturb mode
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables (0) do-not-disturb mode on the VoIP plug-in card.
| Command | Response | Description |
|---|---|---|
| set voip_dnd "VoIP Out" 1 |
val voip_dnd "VoIP Out" 1 |
Sets all VoIP lines associated wtih the channel "VoIP Out" to Do Not Disturb mode. No incoming calls will be allowed while the do not disturb mode is active. |
| set voip_dnd "VoIP Out" 0 |
val voip_dnd "VoIP Out" 0 |
Turns off do not disturb mode on all VoIP lines associated wtih the channel "VoIP Out" Incoming calls will ring into the room. |
voip_eth_settings VoIP Interface Ethernet Settings
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Default | mode='dhcp' |
| Event Source | No |
This parameter gets or sets the Ethernet settings for the VoIP interface. The system must be rebooted for the Ethernet settings to take effect.
The format of the string is a comma-separated list of name/value pairs with the name and value separated by an equals character, and the value enclosed in single quotes.
The mode attribute is always required. It
must be either dhcp or static.
The addr, dns, gw,
and nm attributes are required when
mode is static and ignored (not
required) when mode is dhcp. They
are always returned in the acknowledgement. For the
dhcp case, they specify the values obtained from
the DHCP server.
The addr attribute specifies the IP address of
the interface. The dns attribute specifies the
domain name server(s). A single server or multiple servers
(separated by spaces) may be specified. The gw
attribute specifies the gateway. The nm
parameter specifies the netmask.
DHCP Example
set voip_eth_settings "VoIP In" "mode='dhcp'"
val voip_eth_settings "VoIP In" "mode='dhcp',addr='172.22.2.129',dns='172.22.1.1 172.22.1.2',gw='172.22.2.254',nm='255.255.255.0'"
Static IP Example
set voip_eth_settings "VoIP In" "mode='static',addr='172.22.2.200',dns='172.22.1.1',gw='172.22.2.254',nm='255.255.255.0'"
val voip_eth_settings "VoIP In" "mode='static',addr='172.22.2.200',dns='172.22.1.1',gw='172.22.2.254',nm='255.255.255.0'"
voip_eth_vlan_id Set VLAN ID for VoIP card
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Minimum: -1, Maximum: 4096 |
| Default | -1 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the VLAN ID for the VoIP card. A value of -1 corresponds to "disabled."
voip_factory_reset Reset VoIP Plug-In Card to Factory State
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
This parameter resets the VoIP plug-in card to its factory state.
| Command | Response | Description |
|---|---|---|
| set voip_factory_reset "VoIP In" |
val voip_factory_reset "VoIP In" |
Resets the SoundStructure VoIP Interface to a factory fresh state. This will clear all settings in the SoundStructure VoIP Interface. |
voip_forward Forward a Call
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used to forward a call.
voip_hold Place Current Call on Hold
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter places the current call on hold.
| Command | Response | Description |
|---|---|---|
| set voip_hold "VoIP Out" |
val voip_hold "VoIP Out" |
Places the call appearance on the currently selected line associated with the channel "VoIP Out" into a Hold state. To remove the hold state, use the voip_resume command. There are additional system acknowledgments that are generated when the call is placed on hold including voip_call_appearance_info, voip_call_appearance_state (set to "ncas_call_hold"), voip_line_state (set to "hold") and phone_connect (set to 0). |
voip_join Add Call to Conference
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used to add a call to the conference.
| Command | Response | Description |
|---|---|---|
| set voip_join "VoIP Out" |
val voip_join "VoIP Out" |
Joins a second call to the first call that has been placed on hold to join the first call with a second call. For more information see the example Dialing two calls on different lines in the SoundStructure Design Guide. |
voip_line Select the Active Line
| Channel Type | Virtual Channel |
| Value Type | Sequence |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| System Limits | Minimum: 1, Maximum: 12 |
| Event Source | No |
This parameter selects the currently active line. Some
parameters (e.g. phone_connect,
phone_dial, phone_redial)
operate on the currently active line, as specified by this
parameter. Setting this parameter is analogous to selecting a
line key on the UI of a Polycom VoIP phone.
voip_line_label Label for Line Key
| Channel Type | Virtual Channel | |||
| Value Type | String | |||
| Read/Write Mode | Read-Only | |||
| Phys Chans | VoIP Telephony Output | |||
| Virt Chans | Mono | |||
| Indices |
|
|||
| System Limits | Max String Length: 128 | |||
| Event Source | No |
This parameter reports the label for the specified line.
voip_line_state State for VoIP Line
| Channel Type | Virtual Channel | ||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Value Type | List | ||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Read/Write Mode | Read-Only | ||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Phys Chans | VoIP Telephony Output | ||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Virt Chans | Mono | ||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Indices |
|
||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Values |
|
||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||
| Event Source | No |
This parameter reports the state for the specified line.
voip_loc_city City Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the city field. This corresponds to the
locInfo.x.A3 Polycom UC Software parameter.
voip_loc_company_name Company Name Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the company name field. This corresponds to the
locInfo.x.NAM Polycom UC Software parameter.
voip_loc_country Country Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the country field. This corresponds to the
locInfo.x.country Polycom UC Software parameter.
voip_loc_description Description Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the description field. This corresponds to the
locInfo.x.label Polycom UC Software parameter.
voip_loc_house_number House Number Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the house number field. This corresponds to the
locInfo.x.HNO Polycom UC Software parameter.
voip_loc_house_number_suffix House Number Suffix Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the house number suffix field. This corresponds to the
locInfo.x.HNS Polycom UC Software parameter.
voip_loc_location Additional Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the additional location field. This corresponds to the
locInfo.x.LOC Polycom UC Software parameter.
voip_loc_postal_code Postal Code Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the postal code field. This corresponds to the
locInfo.x.PC Polycom UC Software parameter.
voip_loc_post_directional Post Directional Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the post directional field. This corresponds to the
locInfo.x.POD Polycom UC Software parameter.
voip_loc_pre_directional Pre Directional Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the pre directional field. This corresponds to the
locInfo.x.PRD Polycom UC Software parameter.
voip_loc_state State Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the state field. This corresponds to the
locInfo.x.A1 Polycom UC Software parameter.
voip_loc_street_name Street Name Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the street name field. This corresponds to the
locInfo.x.RD Polycom UC Software parameter.
voip_loc_street_suffix Street Suffix Location Information
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
When the Lync server is configured with location information, this
parameter displays the street suffix field. This corresponds to the
locInfo.x.STS Polycom UC Software parameter.
voip_local_reset Reset Local Configuration Parameters
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
This parameter resets all local configuration parameters.
voip_message_waiting Indicate Whether Messages are Waiting
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | Yes |
This parameter indicates whether or not a voice mail message is waiting for the VoIP interface.
voip_net_cfg_save Save VoIP Network Settings
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
This parameter causes the VoIP network settings to be written to the flash on the VoIP card.
voip_notification VoIP Notification Messages
| Channel Type | Virtual Channel | |||
| Value Type | String | |||
| Read/Write Mode | Read-Only | |||
| Phys Chans | VoIP Telephony Input | |||
| Virt Chans | Mono | |||
| Indices |
|
|||
| Event Source | No |
The VoIP interface may need to display notifications to the user; for example, to indicate that the network connection is down. This parameter provides access to a queue of messages from the VoIP interface. There is a maximum of 32 messages in the queue, but typically there are few or no messages to display. The messages are added to the queue starting with the first index (1), then continuing with the next higher index. All messages after the last queue entry are reported as blank (an empty string).
voip_popup VoIP Popup Message
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| Event Source | No |
The VoIP interface may need to display immediate notifications to the user. This parameter is used by the system to send those notifications. An automatic status message is generated with this parameter whenever the VoIP interface needs to display an immediate notification. Querying this parameter will return the most recent popup message.
voip_prov_serv_address Set provisioning server address for VoIP card
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Default | |
| Event Source | No |
This parameter sets the address of the provisioning server for the VoIP interface.
voip_prov_serv_password Set password for VoIP card provisioning server
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Default | |
| Event Source | No |
This parameter sets the password for the provisioning server used by the VoIP interface.
voip_prov_serv_type Set provisioning server type for VoIP card
| Channel Type | Virtual Channel | |||||||||||||||
| Value Type | List | |||||||||||||||
| Read/Write Mode | Read/Write | |||||||||||||||
| Phys Chans | VoIP Telephony Input | |||||||||||||||
| Virt Chans | Mono | |||||||||||||||
| Values |
|
|||||||||||||||
| Event Source | No |
This parameter controls the provisioning server type for the VoIP interface.
voip_prov_serv_user Set username for VoIP card provisioning server
| Channel Type | Virtual Channel |
| Value Type | String |
| Read/Write Mode | Read/Write |
| Phys Chans | VoIP Telephony Input |
| Virt Chans | Mono |
| System Limits | Max String Length: 256 |
| Default | |
| Event Source | No |
This parameter sets the username for the provisioning server used by the VoIP interface.
voip_reboot Reboot VoIP Plug-in Card
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter causes the VoIP Plug-in Card to reboot.
| Command | Response | Description |
|---|---|---|
| set voip_reboot "VoIP Out" |
val voip_reboot "VoIP Out" |
Reboots the SoundStructure VoIP Interface that is associated with the output channel name "VoIP Out". The SoundStructure system is not rebooted, only the SoundStructure VoIP Interface. |
voip_resume Resume a Call That is On Hold
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter takes the "active call" out of hold.
| Command | Response | Description |
|---|---|---|
| set voip_resume "VoIP Out" |
val voip_resume "VoIP Out" |
Resumes a call that has been placed on hold with the voip_hold command for the channel "VoIP Out". There are additional system acknowledgments that are generated when the call is resumed including voip_call_appearance_info, voip_call_appearance_state (set to "proceeding" and then to "connected"), voip_line_state (set to "proceed" and then "active_hd") and phone_connect (restored to 1). |
voip_send Send Call That Can't be Auto-Dialed
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
Setting this parameter causes a call to be placed with the digits dialed so far.
voip_split Add Call to Conference
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter splits all calls in a conference into individual calls on hold.
voip_status Query Status of VoIP Card
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read-Only | ||||||
| Phys Chans | VoIP Telephony Output | ||||||
| Virt Chans | Mono | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter indicates the status of the VoIP plug-in
card. The three values correspond to the state of the status
LED on the VoIP plug-in card as follows: ok =
solid, booting = flashing.
voip_transfer Transfer a Call
| Channel Type | Virtual Channel |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Phys Chans | VoIP Telephony Output |
| Virt Chans | Mono |
| Event Source | No |
This parameter is used to transfer a call.
| Command | Response | Description |
|---|---|---|
| set voip_transfer "VoIP Out" |
val voip_transfer "VoIP Out" |
This command is used to initiate a call transfer of the active call appearance on the "VoIP" channel. This result of sending this command will be to generate dialtone to supporting dialing a second number. Once the second number is connected, you send a second voip_transfer command to complete the transfer. See the example in the SoundStructure VoIP Interface chapter in the SoundStructure Design Guide. |
The line outputs (both conferencing and sound reinforcement) each have a graphic eq that can have either 10 bands (1 octave), 15 bands (2/3 octave), or 31 bands (1/3 octave).
All physical channel types except the signal generator support a high-pass filter, a low-pass filter, a high-shelf filter, and a low-shelf filter.
The line out physical channel types (both conferencing and sound reinforcement) support a horn eq.
eq_en Enable All Equalizer Processing
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) all equalizer processing (peq, geq, etc.) for
the specified virtual channel.
| Command | Response | Description |
|---|---|---|
| set eq_en "Amplifier" 1 |
val eq_en "Amplifier" 1 |
Enables the equalization processing for the channel "Amplifier". |
eq_type Select Graphic or Parametric Equalizer
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output | ||||||
| Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
The line outputs may have either a graphic or parametric equalizer. This parameter selects which will be used for a given virtual channel.
geq_compensate Enable Gain Compensation for Graphic Equalizer
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) gain compensation for the graphic
equalizer.
geq_en Enable Graphic Equalizer
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the graphic equalizer.
geq_gain Gain of Graphic Equalizer Band
| Channel Type | Virtual Channel | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.01 | |||
| Default | 0.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
Set the gain of the specified band in the graphic equalizer. The index must be between 1 and 10 for 1 octave eq, between 1 and 15 for 2/3 octave eq, and between 1 and 31 for 1/3 octave eq.
geq_type Graphic Equalizer Type
| Channel Type | Virtual Channel | |||||||||
| Value Type | List | |||||||||
| Read/Write Mode | Read/Write | |||||||||
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output | |||||||||
| Virt Chans | Mono, Stereo | |||||||||
| Values |
|
|||||||||
| Event Source | No |
This parameter sets the type of the graphic equalizer.
high_shelf_en Enable High Shelving Filter
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the high shelving filter for the specified
virtual channel.
high_shelf_frequency Frequency of High Shelving Filter
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 500.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the cutoff frequency (in Hz) of the high shelving filter. This is the frequency at which the shelving filter's gain is half its maximum gain.
high_shelf_gain Gain of High Shelving Filter
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.01 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) of the high shelving filter at DC.
high_shelf_slope Slope of High Shelving Filter
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | ||||||
| Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter sets the slope of the high shelving filter.
horn_en Enable Horn Equalizer
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the constant directivity horn equalizer for
the specified virtual channel.
horn_frequency Frequency of Horn Equalizer
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 4000.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the cutoff frequency (in Hz) of the constant directivity horn equalizer. This is the frequency above which the gain increases at 6 dB per octave, and below which the gain is 0 dB.
hpf_en Enable High-Pass Filter
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the high-pass filter for the specified
virtual channel.
hpf_frequency Frequency of High-Pass Filter
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 160.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the frequency (in Hz) of the high-pass filter. For Butterworth filters this is the 3 dB frequency, but for Linkwitz-Riley filters, this is the 6 dB frequency.
hpf_order Order of High-Pass Filter
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 8 |
| Default | 2 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the order of the high-pass filter. Linkwitz-Riley filters only support even orders. If an odd order is specified for a Linkwitz-Riley filter, it will be internally rounded up to an even number.
hpf_type Type of High-Pass Filter
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | ||||||
| Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter sets the type of analog filter prototype used for the high-pass filter.
low_shelf_en Enable Low Shelving Filter
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the low shelving filter for the specified
virtual channel.
low_shelf_frequency Frequency of Low Shelving Filter
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 500.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the cutoff frequency (in Hz) of the low shelving filter. This is the frequency at which the shelving filter's gain is half its maximum gain.
low_shelf_gain Gain of Low Shelving Filter
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.01 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain (in dB) of the low shelving filter at DC.
low_shelf_slope Slope of Low Shelving Filter
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | ||||||
| Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter sets the slope of the low shelving filter.
lpf_en Enable Low-Pass Filter
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the low-pass filter for the specified
virtual channel.
lpf_frequency Frequency of Low-Pass Filter
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 16000.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the frequency (in Hz) of the low-pass filter. For Butterworth filters this is the 3 dB frequency, but for Linkwitz-Riley filters, this is the 6 dB frequency.
lpf_order Order of Low-Pass Filter
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 8 |
| Default | 2 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the order of the low-pass filter. Linkwitz-Riley filters only support even orders. If an odd order is specified for a Linkwitz-Riley filter, it will be internally rounded up to an even number.
lpf_type Type of Low-Pass Filter
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | ||||||
| Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter sets the type of analog filter prototype used for the low-pass filter.
peq_band_en Enable Parametric Equalizer Band
| Channel Type | Virtual Channel | |||
| Value Type | Boolean | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| Default | 0 | |||
| Event Source | No |
This parameter enables (1) or disables
(0) the specified band of the parametric
equalizer for the specified virtual channel. For conference
link physical channels, the band index must be between 1 and
5. For all other physical channels, the band index must be
between 1 and 10.
peq_bandwidth Bandwidth of Parametric Equalizer Band
| Channel Type | Virtual Channel | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| System Limits | Minimum: 0.05, Maximum: 2.0, Resolution: 0.01 | |||
| Default | 0.5 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter sets the bandwidth (in octaves) of the specified parametric equalizer band. In the case of peaking filters, this is the bandwidth at which the gain is half the peak gain (in dB). For notch filters, this is the 3 dB bandwidth. For all-pass filters, this is the bandwidth at which the phase shift is +/- 90 degrees. For conference link physical channels, the band index must be between 1 and 5. For all other physical channels, the band index must be between 1 and 10.
peq_frequency Frequency of Parametric Equalizer Band
| Channel Type | Virtual Channel | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 | |||
| Default | 1000.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter sets the frequency (in Hz) of the specified parametric equalizer band. In the case of peaking and notch types, this is the frequency at which the filter applies maximum (or minimum) gain. For all-pass filters, this is the frequency at which the phase shift is 180 degrees. For conference link physical channels, the band index must be between 1 and 5. For all other physical channels, the band index must be between 1 and 10.
peq_gain Gain of Parametric Equalizer Band
| Channel Type | Virtual Channel | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.01 | |||
| Default | 0.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter sets the gain (in dB) of the specified parametric equalizer band. For conference link physical channels, the band index must be between 1 and 5. For all other physical channels, the band index must be between 1 and 10.
peq_type Type of Parametric Equalizer Band
| Channel Type | Virtual Channel | |||||||||
| Value Type | List | |||||||||
| Read/Write Mode | Read/Write | |||||||||
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output | |||||||||
| Virt Chans | Mono, Stereo | |||||||||
| Indices |
|
|||||||||
| Values |
|
|||||||||
| Event Source | No |
This parameter sets the type of the specified parametric equalizer band. For conference link physical channels, the band index must be between 1 and 5. For all other physical channels, the band index must be between 1 and 10.
Dynamics processing is available on all physical channels except the signal generator and AEC reference. Dynamics processing includes a compressor, limiter, expander, gate, and peak limiter. An additional input gain parameter is provided to change the gain of the signal before the dynamics processor.
dp_en Enable Dynamics Processing
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) all the dynamics processing for the specified
virtual channel.
dp_gate_attack Gate Attack Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 200 |
| Default | 1 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the gate to ramp the gain up to the target gain once the input signal level surpasses the gate threshold.
dp_gate_decay Gate Decay Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 2000 |
| Default | 1000 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the gate to ramp down to the target gain once the input signal drops below the gate threshold and the gate hold time has expired.
dp_gate_en Enable Gate
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the gate function of the dynamics
processor. This parameter and dp_en must be
enabled for the gate to function.
dp_gate_hold Gate Hold Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 2000 |
| Default | 500 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) the input signal level must be below the gate threshold before the gate begins to decay.
dp_gate_ratio Gate Ratio
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1.0, Maximum: 100.0, Resolution: 0.1 |
| Default | 100.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the ratio of the target gain applied by the gate versus the difference between the input signal level and the gate threshold. For example, if the gate ratio is 10 (i.e., 10:1) and the input signal level is 6 dB below the gate threshold, the gate applies -60 dB of gain.
dp_gate_thresh Gate Threshold
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 0.0, Resolution: 0.1 |
| Default | -100.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the RMS level (in dBFS) of the input
signal below which the gate engages. The level must be below
this threshold longer than the gate hold time (set by
dp_gate_hold) before the gate begins to apply
a gain change.
dp_exp_attack Expander Attack Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 200 |
| Default | 10 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the expander to ramp the gain up to the target gain once the input signal level surpasses the expander threshold.
dp_exp_decay Expander Decay Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 2000 |
| Default | 100 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the expander to ramp down to the target gain once the input signal drops below the expander threshold.
dp_exp_en Enable Expander
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the expander function of the dynamics
processor. This parameter and dp_en must be
enabled for the expander to function.
dp_exp_ratio Expander Ratio
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1.0, Maximum: 100.0, Resolution: 0.1 |
| Default | 2.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the ratio of the target gain applied by the expander versus the difference between the input signal level and the expander threshold. For example, if the expander ratio is 2 (i.e., 2:1) and the input signal level is 3 dB below the expander threshold, the expander applies -6 dB of gain.
dp_exp_thresh Expander Threshold
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 0.0, Resolution: 0.1 |
| Default | -100.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the RMS level (in dBFS) of the input signal below which the expander engages.
dp_comp_attack Compressor Attack Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 200 |
| Default | 10 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the compressor to ramp the gain down to the target gain once the input signal level surpasses the compressor threshold.
dp_comp_decay Compressor Decay Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 2000 |
| Default | 100 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the compressor to ramp the gain up to the target gain once the input signal level drops below the compressor threshold.
dp_comp_en Enable Compressor
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the compressor function of the dynamics
processor. This parameter and dp_en must be
enabled for the compressor to function.
dp_comp_ratio Compressor Ratio
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1.0, Maximum: 100.0, Resolution: 0.1 |
| Default | 2.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the ratio of the target gain applied by the compressor versus the difference between compressor threshold and the input signal level. For example, if the compressor ratio is 2 (i.e., 2:1) and the input signal level is 3 dB above the compressor threshold, the compressor applies -1.5 dB of gain.
dp_comp_thresh Compressor Threshold
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 0.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the RMS level (in dBFS) of the input signal above which the compressor engages.
dp_lim_attack Limiter Attack Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 200 |
| Default | 5 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the limiter to ramp the gain down to the target gain once the input signal level surpasses the limiter threshold.
dp_lim_decay Limiter Decay Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 2000 |
| Default | 500 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of time (in milliseconds) it takes the limiter to ramp the gain up to the target gain once the input signal level drops below the limiter threshold.
dp_lim_en Enable Limiter
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the limiter function of the dynamics
processor. This parameter and dp_en must be
enabled for the limiter to function.
dp_lim_ratio Limiter Ratio
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1.0, Maximum: 100.0, Resolution: 0.1 |
| Default | 10.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the ratio of the target gain applied by the limiter versus the difference between the limiter threshold and the input signal level. For example, if the limiter ratio is 10 (i.e., 10:1) and the input signal level is 6 dB above the limiter threshold, the limiter applies -5.4 dB of gain.
dp_lim_thresh Limiter Threshold
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 0.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the RMS level (in dBFS) of the input signal above which the limiter engages.
dp_peak_en Enable Peak Limiter
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the peak limiter function of the dynamics
processor. This parameter and dp_en must be
enabled for the peak limiter to function.
dp_peak_thresh Peak Limiter Threshold
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Input, Analog Telephony Output, VoIP Telephony Input, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 0.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the RMS level (in dBFS) of the input signal above which the peak limiter engages. The peak limiter will ensure that the peak level never exceeds this threshold.
aec_en Enable Acoustic Echo Canceller
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the acoustic echo cancellation (AEC)
algorithm.
aec_noise_fill Enable Noise Fill
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 1 |
| Event Source | No |
This parameter enables (1) or disables
(0) the noise fill algorithm in the AEC.
aec_ref AEC Reference
| Channel Type | Virtual Channel | |||
| Value Type | String | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Telepresence Mic/Line Input | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| System Limits | Max String Length: 256 | |||
| Default | ||||
| Event Source | No |
This parameter is used to set the AEC references for an
given virtual channel. The string argument specifies the name
of the virtual channel that will be the AEC reference. The
string argument must be a valid virtual channel name for a
currently defined virtual channel that is a conferencing line
output (cr_line_out), sound reinforcement line
output (sr_line_out), or submix output
(submix).
The index is used to specify the left (1) or
right (2) reference channels. If neither the
left nor the right channel have references specified, then the
AEC is disabled. If only the left channel is specified, then
the mono AEC algorithm is used. If both the left and right
channel are specified, then the stereo AEC algorithm is
used.
If the reference's virtual channel is mono, then the
corresponding physical channel is used as the AEC reference.
If the reference's virtual channel is stereo, then either the
left or right physical channel is used as a reference,
depending on which index is specified (1 for
left, 2 for right).
If the AEC is on a stereo virtual channel, then the reference specifications apply to both physical channels of the stereo virtual channel.
agc_en Enable Automatic Gain Control
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Telepresence Mic/Line Input, Telepresence Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the automatic gain control (AGC)
algorithm.
agc_max_gain AGC Maximum Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Telepresence Mic/Line Input, Telepresence Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 0.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 6.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the maximum gain (in dB) that can be applied by the AGC.
agc_min_gain AGC Minimum Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Telepresence Mic/Line Input, Telepresence Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 0.0, Resolution: 0.1 |
| Default | -6.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the minimum gain (in dB) that can be applied by the AGC.
delay Amount of Delay
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 0, Maximum: 48000 |
| Default | 0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of delay applied to the audio signal, in samples. The sampling frequency is 48kHz, which means that a sample is 20.83 microseconds. The maximum delay of 96000 samples is equivalent to 2 seconds.
delay_en Enable Signal Delay
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Conferencing Line Output, Sound Reinforcement Mic/Line Input, Sound Reinforcement Line Output, Analog Telephony Output, VoIP Telephony Output, Submix, ConferenceLink Aux Input, ConferenceLink Aux Output, Telepresence Mic/Line Input, Telepresence Line Input, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the delay algorithm.
fb_en Enable Feedback Reduction
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the feedback reduction algorithm.
fb_filter_bandwidth Feedback Reduction Filter Bandwidth
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 0.03, Maximum: 1.0, Resolution: 0.01 |
| Default | 0.1 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the bandwidth (in octaves) for all the filters of the feedback reduction algorithm.
fb_filter_decay_en Enable Filter Decay Mode in Feedback Reduction Algorithm
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) filter decay mode for the feedback reduction
algorithm. If filter decay mode is enabled, the adaptive
notch filters can slowly decay to 0 dB if no singing is
detected at that frequency. This mode is useful in rooms with
high noise or where there is a lot of motion.
fb_filter_reset Reset One of the Feedback Reduction Filters
| Channel Type | Virtual Channel | |||
| Value Type | Void | |||
| Read/Write Mode | Write-Only | |||
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| Event Source | No |
Setting this parameter resets the specified filter in the feedback reduction algorithm. Redpoint will likely set this parameter for filters it has converted to fixed parametric EQ filters.
fb_filter_max_depth Maximum Filter Depth for Feedback Reduction Filters
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -100.0, Maximum: 0.0, Resolution: 0.1 |
| Default | -15.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the maximum attenuation (in dB) that can be applied for any feedback reduction filter.
fb_safe_mode_atten Safe Mode Attenuation for Feedback Reduction
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 0.0, Maximum: 100.0, Resolution: 0.1 |
| Default | 3.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter defines the maximum amount of attenuation (in dB) applied to the input if all the filters are used up and the feedback reduction algorithm continues to detect singing. Setting this parameter to 0 dB means that no attenuation is performed even if all the filters are used up.
mic_source_index Select Mic Audio Source Index
| Channel Type | Virtual Channel | |||
| Value Type | Sequence | |||
| Read/Write Mode | Read/Write | |||
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input, Telepresence Line Input | |||
| Virt Chans | Mono, Stereo | |||
| Indices |
|
|||
| System Limits | Minimum: 1, Maximum: 15 | |||
| Default | 1 | |||
| Event Source | No |
This parameter selects the index of the audio source for
the corresponding cr_mic_in or
sr_mic_in physical channels. The index required
for this parameter indicates to which physical channel of the
virtual channel this parameter will be applied. For example,
indices 1 and 2 correspond to the
left and right physical channels of a stereo virtual
channel.
The value of this parameter indicates the index of the
audio source type (mic_source_type) that will
be routed to the physical channel.
When mic_source_type is
analog, this parameter has no effect.
When mic_source_type is
clink_mic, the value of this parameter indicates
which ConferenceLink mic element will be routed to the
physical channel.
For example, assume a virtual channel has been defined like this:
vcdef "Stereo Clink Mic" stereo cr_mic_in 1 2
And the source type has been set to clink_mic
like this:
set mic_source_type "Stereo Clink Mic" clink_mic
Then we issue these commands:
set mic_source_index "Stereo Clink Mic" 1 5
set mic_source_index "Stereo Clink Mic" 2 6
These commands set the left and right channels of the "Stereo Clink Mic" virtual channel to use the 2nd and 3rd elements of the 2nd ConferenceLink mic.
| Command | Response | Description |
|---|---|---|
| get mic_source_index "Ceiling Mic 1 B" 1 |
val mic_source_index "Ceiling Mic 1 B" 1 2 |
Queries which microphone index of the microphone array is associated with the virtual channel "Ceiling Mic 1 B". Since "Ceiling Mic 1 B" is the second element of the first digital array microphone, the value 2 is returned. |
| get mic_source_index "Ceiling Mic 2 A" 1 |
val mic_source_index "Ceiling Mic 2 A" 1 4 |
Queries which microphone index of the microphone array is associated with the virtual channel "Ceiling Mic 2 A". Since "Ceiling Mic 2 A" is the first element of the second digital array microphone, the value 4 is returned. |
| get mic_source_index "Ceiling Mic 2 B" 1 |
val mic_source_index "Ceiling Mic 2 B" 1 5 |
Queries which microphone index of the microphone array is associated with the virtual channel "Ceiling Mic 2 B".Since "Ceiling Mic 2 B" is the second element of the second digital array microphone, the value 5 (=3+2) is returned. |
mic_source_type Select Mic Audio Source Type
| Channel Type | Virtual Channel | |||||||||
| Value Type | List | |||||||||
| Read/Write Mode | Read/Write | |||||||||
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input, Telepresence Line Input | |||||||||
| Virt Chans | Mono, Stereo | |||||||||
| Values |
|
|||||||||
| Event Source | No |
This parameter selects the audio source for the
corresponding cr_mic_in or sr_mic_in
physical channels. The analog type selects
the analog microphone audio. The clink_mic type
selects one of the ConferenceLink microphone elements.
Control of which element is selected is done through the
mic_source_index parameter.
The submix type selects a submix from the local
device to use microphone audio processing.
The physical channel of the submix to be used is determined
by the mic_source_index parameter.
The submix option is for experimental use only.
| Command | Response | Description |
|---|---|---|
| set mic_source_type "Table Mic 1" analog |
val mic_source_type "Table Mic 1" analog |
Sets the mic_source_type for "Table Mic 1" to analog. |
| get mic_source_type "Ceiling Mic 1 A" |
val mic_source_type "Ceiling Mic 1 A" clink_mic |
Queries the mic_source_type for "Ceiling Mic 1 A". |
nc_en Enable Noise Canceller
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) the noise cancellation (NC)
algorithm.
nc_level Noise Cancellation Level
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Analog Telephony Input, VoIP Telephony Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 0.0, Maximum: 20.0, Resolution: 1.0 |
| Default | 10.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the amount of cancellation (in dB) applied by the noise cancellation algorithm.
sig_gen_gain Signal Generator Gain
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Signal Generator |
| Virt Chans | Mono |
| System Limits | Minimum: -100.0, Maximum: 20.0, Resolution: 0.1 |
| Default | -30.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the gain of the signal produced by the signal generator, in dB. A gain of 0 dB will produce a signal that has the same RMS level as a sine wave at -20 dBFS (the nominal signal level). This means that not all signal types will have the same peak level, and some types may clip before a gain of 20 dB is applied.
sig_gen_sweep_start Signal Generator Sweep Start Frequency
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Signal Generator |
| Virt Chans | Mono |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 20.0 |
| User Limits Supported | No |
| Event Source | No |
When the signal generator's sig_gen_type is
set to sweep, this parameter sets the frequency
(in Hz) at which the sweep generator begins. The direction of
the frequency sweep will be up or down depending on whether
this parameter is higher or lower than the
sig_gen_sweep_stop parameter.
sig_gen_sweep_step Signal Generator Sweep Step Size
| Channel Type | Virtual Channel | ||||||||||||||||||
| Value Type | List | ||||||||||||||||||
| Read/Write Mode | Read/Write | ||||||||||||||||||
| Phys Chans | Signal Generator | ||||||||||||||||||
| Virt Chans | Mono | ||||||||||||||||||
| Values |
|
||||||||||||||||||
| Event Source | No |
When the signal generator's sig_gen_type is
set to sweep, this parameter sets the step size
of the sweep generator. The signal generator can sweep
continuously, in fractional octave steps, or in full octave
steps.
sig_gen_sweep_stop Signal Generator Sweep Stop Frequency
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Signal Generator |
| Virt Chans | Mono |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 20000.0 |
| User Limits Supported | No |
| Event Source | No |
When the signal generator's sig_gen_type is
set to sweep, this parameter sets the frequency
(in Hz) at which the sweep generator stops. The direction of
the frequency sweep will be up or down depending on whether
the sig_gen_sweep_start parameter is higher or
lower than this parameter.
sig_gen_sweep_time Signal Generator Sweep Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Signal Generator |
| Virt Chans | Mono |
| System Limits | Minimum: 10, Maximum: 60000 |
| Default | 10000 |
| User Limits Supported | No |
| Event Source | No |
When the signal generator's sig_gen_type is
set to sweep, this parameter sets the duration
(in milliseconds) that the sweep generator takes to sweep from
its start frequency to its stop frequency.
sig_gen_tone_freq Signal Generator Tone Frequency
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Signal Generator |
| Virt Chans | Mono |
| System Limits | Minimum: 20.0, Maximum: 20000.0, Resolution: 0.1 |
| Default | 1000.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the frequency (in Hz) of the sine wave
produced by the signal generator when its
sig_gen_type is set to tone.
sig_gen_type Signal Generator Type
| Channel Type | Virtual Channel | ||||||||||||
| Value Type | List | ||||||||||||
| Read/Write Mode | Read/Write | ||||||||||||
| Phys Chans | Signal Generator | ||||||||||||
| Virt Chans | Mono | ||||||||||||
| Values |
|
||||||||||||
| Event Source | No |
This parameter sets the type of signal produced by the
signal generator. The options are pink noise
(pink), white noise (white), a sine
wave at a single frequency (tone), and a sine
wave swept across a range of frequencies
(sweep).
smartpairing_en Enable SmartPairing
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables (1) or disables
(0) SmartPairing.
smartpairing_dev_id Set SmartPairing Source
| Channel Type | Virtual Channel |
| Value Type | Sequence |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Line Output, Sound Reinforcement Line Output, Telepresence Line Output |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 8 |
| Default | 1 |
| Event Source | No |
This parameter sets the device that is connected to the Polycom HDX Series video code or Polycom RealPresence Group Series video codec that is acting as the SmartPairing signal source for this system. That signal will be rendered on this output.
cr_ungated_type Select Processing for Ungated Signal
| Channel Type | Virtual Channel | ||||||||||||
| Value Type | List | ||||||||||||
| Read/Write Mode | Read/Write | ||||||||||||
| Phys Chans | Conferencing Mic/Line Input, Telepresence Mic/Line Input | ||||||||||||
| Virt Chans | Mono, Stereo | ||||||||||||
| Values |
|
||||||||||||
| Event Source | No |
This parameter selects the version of signal to use for the ungated triune signal of the specified virtual channel.
sr_delay_type Select Delay for Sound Reinforcement Signal
| Channel Type | Virtual Channel | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input | ||||||
| Virt Chans | Mono, Stereo | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter selects the version of signal to use for the sound reinforcement triune signal of the specified virtual channel.
sr_ungated_type Select Processing for Ungated Signal
| Channel Type | Virtual Channel | ||||||||||||
| Value Type | List | ||||||||||||
| Read/Write Mode | Read/Write | ||||||||||||
| Phys Chans | Sound Reinforcement Mic/Line Input | ||||||||||||
| Virt Chans | Mono, Stereo | ||||||||||||
| Values |
|
||||||||||||
| Event Source | No |
This parameter selects the version of signal to use for the ungated triune signal of the specified virtual channel.
ungated_delay_comp_en Enable Delay Compensation for Triune Signals
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
Delay compensation only applies to the ungated signal, and only when it is using the line or bypass options.
am_adapt_thresh Automixer Adaptive Threshold
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 0.0, Maximum: 100.0, Resolution: 0.1 |
| Default | 10.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter defines how much louder (in dB) the microphone's signal level must be above its measured noise floor before it is eligible to be considered active. Higher settings will make the microphone's gating less sensitive, while lower settings will make it more sensitive.
am_camera_activity_time Automixer Camera Activity Time
| Channel Type | Global System | |||
| Value Type | Integer | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 10, Maximum: 10000 | |||
| Default | 2000 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter defines the amount of time (in ms) a signal must be active before showing up on the camera activity meter. In general, it should be set somewhat longer than the hold time of the automixer.
am_chairman Automixer Chairman Microphone
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
When this parameter is set to 1, the
microphone is considered a chairman microphone.
am_chan_bias Automixer Channel Bias
| Channel Type | Virtual Channel |
| Value Type | Floating-Point |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.1 |
| Default | 0.0 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the channel bias (in dB) for the associated microphone. For the purpose of determining activity status, this microphone is treated as though its level were higher or lower (according to the setting of the parameter) than its actual measured level.
am_decay_time Automixer Decay Time
| Channel Type | Global System | |||
| Value Type | Integer | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 10, Maximum: 10000 | |||
| Default | 1000 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter defines how long (in ms) the gain of a gated microphone in the specified automixer group takes to transition between fully open and its off attenuation value when it is time for the microphone to gate off.
This parameter is only used if the automixer group is in gating mode. If it is gain sharing mode, the parameter is ignored.
am_en Enable Automixer
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
This parameter enables or disables the automixer for the virtual channel. When disabled, the microphone is completely pulled out of the automixer (so that it does not affect the gating of other channels), and a gain of 0 dB is applied to the channel (so that it is always open).
| Command | Response | Description |
|---|---|---|
| set am_en "Table Mic 1" 1 |
val am_en "Table Mic 1" 1 |
Enables the automixer processing for the channel "Table Mic 1". |
am_gain_sharing Enable Gain-Sharing Automixer Mode
| Channel Type | Global System | |||
| Value Type | Boolean | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| Default | 0 | |||
| Event Source | No |
This parameter selects gain-sharing mode for the specified
automixer group when set to 1. Otherwise, the
microphones in the automixer group are in gating mode.
| Command | Response | Description |
|---|---|---|
| set am_gain_sharing 2 1 |
val am_gain_sharing 2 1 |
Selects the gain sharing automixer forthe microphones in automixer group 2. |
| set am_gain_sharing 1 0 |
val am_gain_sharing 1 0 |
Selects the gated automixer for the microphones in automixer group 1. |
am_group Automixer Group
| Channel Type | Virtual Channel |
| Value Type | Sequence |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 63 |
| Default | 1 |
| Event Source | No |
This parameter selects the automixer group in which the microphone is a member.
| Command | Response | Description |
|---|---|---|
| set am_group "Table Mic 1" 1 |
val am_group "Table Mic 1" 1 |
Assigns "Table Mic 1" to automixer group 1. |
am_hold_time Automixer Hold Time
| Channel Type | Global System | |||
| Value Type | Integer | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 100, Maximum: 10000 | |||
| Default | 500 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter defines how long (in ms) the microphone in the specified automixer group will be considered active after the last detected significant level on the microphone.
am_last_mic_mode Automixer Last Mic Mode
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| Default | 0 |
| Event Source | No |
When this parameter is set to 1, the last mic
mode is enabled on the microphone.
am_nom_limit NOM Limit
| Channel Type | Global System | |||
| Value Type | Integer | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 0, Maximum: 128 | |||
| Default | 16 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter sets the NOM limit for the microphone with respect to its automixer group.
am_off_atten Automixer Off Attenuation
| Channel Type | Global System | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 0.0, Maximum: 100.0, Resolution: 0.1 | |||
| Default | 15.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter defines how much attenuation (in dB) is applied to a gated microphone in the specified group when the microphone is fully gated off.
This parameter is only used if the automixer group is in gating mode. If it is gain sharing mode, the parameter is ignored.
am_priority Automixer Microphone Priority
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Conferencing Mic/Line Input, Sound Reinforcement Mic/Line Input, Telepresence Mic/Line Input |
| Virt Chans | Mono, Stereo |
| System Limits | Minimum: 1, Maximum: 4 |
| Default | 1 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the priority of the microphone. A priority of 1 is the highest priority (most favored), while a priority of 4 is the lowest priority (least favored).
am_priority_atten Automixer Priority Attenuation
| Channel Type | Global System | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 0.0, Maximum: 100.0, Resolution: 0.1 | |||
| Default | 0.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter sets the amount of attenuation (in dB) that is applied to the microphones in the specified automixer group if a higher priority microphone in the group is currently active.
am_slope Gain Sharing Automixer Slope
| Channel Type | Global System | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read/Write | |||
| Indices |
|
|||
| System Limits | Minimum: 0.0, Maximum: 10.0, Resolution: 0.1 | |||
| Default | 2.0 | |||
| User Limits Supported | No | |||
| Event Source | No |
This parameter defines how much attenuation (in dB) is applied to microphones in the specified automixer group when they don't have the highest level in the group. For example, if a microphone has a level that is 6.0 dB lower than the loudest mic, and its slope is 2.0, then 12.0 dB of attenuation will be applied to the microphone.
This parameter is only used if the automixer group is in gain sharing mode. If the automixer group is in gating mode, the parameter is ignored.
analog_gpio_value Analog GPIO Value
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Analog General Purpose I/O Input |
| Virt Chans | Control |
| System Limits | Minimum: 0, Maximum: 255 |
| Default | 0 |
| User Limits Supported | No |
| Event Source | Yes |
This parameter gets or sets the value of the analog gpio pin. Writing an input has no effect and returns the current value of the input.
| Command | Response | Description |
|---|---|---|
| get analog_gpio_value "Analog Logic Pin" |
val analog_gpio_value "Analog Logic Pin" 0 |
Returns the analog voltage associated with the analog logic pin "Analog Logic Pin". The logic pin "Analog Logic Pin" must have been created with a vcdef command. |
digital_gpio_held Digital GPIO Pin Held Status
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read-Only |
| Phys Chans | Digital General Purpose I/O Input |
| Virt Chans | Control |
| Event Source | Yes |
This parameter indicates when a digital input pin is held.
It is similar in functionality to the
ir_key_held parameter. The hold time and
repeat time are specified by the
digital_gpio_hold_time and
digital_gpio_repeat_time parameters. When the
pin is held for the hold time, a status message will be
generated. If the pin remains held, status messages will be
generated with a period equal to the repeat time.
| Command | Response | Description |
|---|---|---|
| get digital_gpio_held "Button 1" |
val digital_gpio_held "Button 1" 0 |
Queries whether the logic input "Button 1" has been held for at least the digital_gpio_hold_time. If the digital_gpio_hold_time is set to 0, then digital_gpio_held will always return 0. |
digital_gpio_hold_time Digital GPIO Pin Hold Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Digital General Purpose I/O Input |
| Virt Chans | Control |
| System Limits | Minimum: 0, Maximum: 10000 |
| Default | 0 |
| User Limits Supported | No |
| Event Source | No |
This parameter specifies the amount of time (in
milliseconds) that a GPIO pin must be held for the first
digital_gpio_held status message to be sent.
Setting this parameter to 0 indicates that
digital_gpio_held messages will not be
generated.
| Command | Response | Description |
|---|---|---|
| set digital_gpio_hold_time "Button 1" 1000 |
val digital_gpio_hold_time "Button 1" 1000 |
Sets the hold time for logic input "Button 1" to 1000msec. If the switch has been held closed(0) for this amount of time, a digital_gpio_held message will be sent with the value 1. |
digital_gpio_repeat_time Digital GPIO Pin Repeat Time
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Digital General Purpose I/O Input |
| Virt Chans | Control |
| System Limits | Minimum: 1, Maximum: 10000 |
| Default | 1000 |
| User Limits Supported | No |
| Event Source | No |
This parameter specifies the amount of time (in
milliseconds) between digital_gpio_held
status messages when a GPIO pin is continually held.
| Command | Response | Description |
|---|---|---|
| set digital_gpio_repeat_time "Button 1" 2000 |
val digital_gpio_repeat_time "Button 1" 2000 |
Sets the repeat time for logic input "Button 1" to 2000msec causing the digital_gpio_held messages to be spaced 2000msec apart if the switch is held closed. |
digital_gpio_state Digital GPIO Pin Status
| Channel Type | Virtual Channel |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Phys Chans | Digital General Purpose I/O Input, Digital General Purpose I/O Output |
| Virt Chans | Control |
| Default | 0 |
| Event Source | Yes |
This parameter gets or sets the value of the digital gpio pin. Writing an input has no effect and returns the current value of the input.
| Command | Response | Description |
|---|---|---|
| set digital_gpio_state "LED 5" 1 |
val digital_gpio_state "LED 5" 1 |
Sets the logic output pin "LED 5" to 1 which will enable the logic output to drive an LED that may be connected to that logic output channel. |
| get digital_gpio_state "Button 3" |
val digital_gpio_state "Button 3" 1 |
Queries the logic input pin "Button 3" to see if the switch is open (1) or closed (0). |
digital_gpio_value Digital GPIO Array Value
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Digital General Purpose I/O Input, Digital General Purpose I/O Output |
| Virt Chans | Control Array |
| System Limits | Minimum: 0, Maximum: 2147483647 |
| Default | 0 |
| User Limits Supported | No |
| Event Source | Yes |
This parameter gets or sets the value of the digital gpio array. Writing an input has no effect and returns the current value of the input.
| Command | Response | Description |
|---|---|---|
| set digital_gpio_value "Logic Out Array" 7 |
val digital_gpio_value "Logic Out Array" 7 |
Sets the value of the logic output array "Logic Out Array" to 7 which turns on the least significant 3 logic ouput pins of this array to 1. |
| get digital_gpio_value "Logic Input Array" |
val digital_gpio_value "Logic Input Array" 4 |
Queries the logic input array "Logic Input Array" to see which pins in the array are open (1) or closed (0). |
auth_password Set Authentication Password
| Channel Type | Global System |
| Value Type | String |
| Read/Write Mode | Write-Only |
| System Limits | Max String Length: 128 |
| Event Source | No |
This parameter sets the authentication password. The default value for this parameter is "456".
| Command | Response | Description |
|---|---|---|
| set auth_password "12345" |
val auth_password "****" |
Sets the authentication password to "12345". The acknowledgment masks the password and returns "****" instead of the password. |
clink_call_active CLink2 Call Active Status
| Channel Type | Device-Specific System |
| Value Type | Boolean |
| Read/Write Mode | Read-Only |
| Event Source | Yes |
This parameter gets the call active status of a Polycom HDX Series video codec or Polycom RealPresence Group Series video codec. This is true whenever the codec has an active video or voice call. A status message is generated whenever the call active status is changed by a Polycom video codec.
| Command | Response | Description |
|---|---|---|
| get clink_call_active 2 |
val clink_call_active 2 1 |
Queries the state of the clink_call_active parameter for SoundStructure device 2. In this example the value 1 was returned which indicates the Polycom video codec is in an active call. |
| get clink_call_active 1 |
val clink_call_active 1 0 |
Queries the state of the clink_call_active parameter for SoundStructure device 1. In this example the value 0 was returned which indicates the Polycom video codec is not in an active call. |
clink_local_call_active CLink2 Local Call Active Status
| Channel Type | Device-Specific System |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| System Limits | Minimum: 0, Maximum: 32 |
| Default | 0 |
| User Limits Supported | No |
| Event Source | Yes |
This parameter gets and sets the call active status that the device
broadcasts to any connected Polycom HDX Series video codec, Polycom
RealPresence Group Series video codec, and/or Polycom microphone
arrays. This primarily controls the state of the green LED on the
Polycom table microphone arrays. Whenever this parameter is a value
greater than 0, the call active status sent to the Polycom
video codec and microphone arrays is set to true. This is implemented
as an integer command so that when a call of interest goes active, it
can be incremented. When the call goes inactive, this parameter can be
decremented. This provides a count of all active calls in the
system.
| Command | Response | Description |
|---|---|---|
| get clink_local_call_active 3 |
val clink_local_call_active 3 1 |
Queries the state of the clink_local_call_active parameter for SoundStructure device 3. In this example the value 1 was returned which indicates that SoundStructure device 3 has 1 active PSTN call in progress. |
| get clink_local_call_active 1 |
val clink_local_call_active 1 0 |
Queries the state of the clink_local_call_active parameter for SoundStructure device 1. In this example the value 0 was returned which indicates the SoundStructure device 1 does not have an active PSTN call in progress. |
clink_mute CLink2 Mute Status
| Channel Type | Device-Specific System |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Default | 0 |
| Event Source | Yes |
This parameter gets or sets the mute status of a Polycom HDX Series video codec, Polycom RealPresence Group Series video codec, and/or Polycom microphone array attached to the indicated device. This does not actually mute any audio. It only reflects the settings of the red mute LEDs on Polycom table microphone arrays, or the mic mute display on a Polycom video codec. A status message is generated whenever the mute status is changed by a Polycom video codec or microphone array.
| Command | Response | Description |
|---|---|---|
| set clink_mute 1 0 |
val clink_mute 1 0 |
Sets the clink_mute of device 1 to 0, i.e., setting clink_mute on device 1 to false. This will turn off the mute indicator on an Polycom video codec that is attached to this device over conference link. |
| set clink_mute 2 1 |
val clink_mute 2 1 |
Sets the clink_mute for SoundStructure device 2 to be 1, or true. |
clink_num_video_channels CLink2 Call Active Status
| Channel Type | Device-Specific System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| Event Source | Yes |
This parameter gets the number of channels present in the current video call of a Polycom HDX Series video codec or Polycom RealPresence Group Series video codec. If there is no video call in progress, the number of channels from the most recent video call is reported. A status message is generated whenever the number of channels is changed by a Polycom video codec (at the beginning of a new call).
clink_volume CLink2 Volume Status
| Channel Type | Device-Specific System |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| System Limits | Minimum: 0, Maximum: 51 |
| Default | 31 |
| User Limits Supported | No |
| Event Source | Yes |
This parameter gets or sets the volume status of a Polycom HDX Series video codec or Polycom RealPresence Group Series video codec. This does not actually adjust any gains. It only reflects the settings of the on-screen volume control bar of the video codec. A status message is generated whenever the volume is changed by a Polycom video codec.
| Command | Response | Description |
|---|---|---|
| get clink_volume 1 |
val clink_volume 1 31 |
Queries the state of the clink_volume parameter for SoundStructure device 1 and returns the current value of 31 as the volume. |
eth_auth_mode Ethernet Authentication Mode
| Channel Type | Global System | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter selects the port for Ethernet control
connections. If it is set to open, then
connections are accepted on port 52774 and do not require
authentication. If it is set to auth, then
connections are accepted on port 52775 and authentication is
required.
eth_settings Ethernet Settings
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read/Write |
| System Limits | Max String Length: 256 |
| Default | mode='dhcp' |
| Event Source | No |
This parameter gets or sets the Ethernet settings. When this parameter is set, the Ethernet interface is automatically restarted with the new settings.
The format of the string is a comma-separated list of name/value pairs with the name and value separated by an equals character, and the value enclosed in single quotes.
The mode attribute is always required. It
must be either dhcp or static.
The addr, dns, gw,
and nm attributes are required when
mode is static and ignored (not
required) when mode is dhcp. They
are always returned in the acknowledgement. For the
dhcp case, they specify the values obtained from
the DHCP server.
The addr attribute specifies the IP address of
the interface. The dns attribute specifies the
domain name server(s). A single server or multiple servers
(separated by spaces) may be specified. The gw
attribute specifies the gateway. The nm
parameter specifies the netmask.
DHCP Example
set eth_settings 1 "mode='dhcp'"
val eth_settings 1 "mode='dhcp',addr='172.22.2.129',dns='172.22.1.1 172.22.1.2',gw='172.22.2.254',nm='255.255.255.0'"
Static IP Example
set eth_settings 1 "mode='static',addr='172.22.2.200',dns='172.22.1.1',gw='172.22.2.254',nm='255.255.255.0'"
val eth_settings 1 "mode='static',addr='172.22.2.200',dns='172.22.1.1',gw='172.22.2.254',nm='255.255.255.0'"
| Command | Response | Description |
|---|---|---|
| set eth_settings 1 "mode='dhcp'" |
val eth_settings 1 "mode='dhcp',addr='172.22.2.129',dns='172.22.1.1 172.22.1.2',gw='172.22.2.254',nm='255.255.255.0'" |
Sets the ethernet settings on device 1 to dhcp and returns the full ethernet settings. |
| set eth_settings 1 "mode='static',addr='192.168.10.63',dns='192.168.10.1',gw='192.168.10.254',nm='255.255.255.0'" |
val eth_settings 1 "mode='static',addr='192.168.10.63',dns='192.168.10.1',gw='192.168.10.254',nm='255.255.255.0'" |
Sets the ethernet settings on device 1 to the static IP address of 192.168.10.63 and returns the full ethernet settings. All parameters must be specified even if just changing one parameter. |
| get eth_settings 1 |
val eth_settings 1 "mode='static',addr='192.168.10.63',dns='192.168.10.1',gw='192.168.10.254',nm='255.255.255.0'" |
Queries the ethernet settings on device 1. |
eth_mac Get Ethernet MAC Address
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| System Limits | Max String Length: 17 |
| Event Source | No |
This parameter gets the MAC address for the system's
Ethernet port. The value will be formatted in all lowercase
with bytes separated by a colon, for example
"00:04:f2:bf:00:01".
| Command | Response | Description |
|---|---|---|
| get eth_mac 1 |
val eth_mac 1 "00:04:f2:bf:00:01" |
Queries the ethernet mac address on device 1. |
ir_key_press Key Pressed on IR Remote
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| Phys Chans | Infrared Remote Input |
| Virt Chans | Control |
| System Limits | Minimum: 0, Maximum: 255 |
| Event Source | Yes |
When queried, this parameter returns the keycode value of the last key that was pressed on the IR remote. As an event, a status message is generated whenever a key is pressed on the IR remote.
ir_key_held Key Held on IR Remote
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| Phys Chans | Infrared Remote Input |
| Virt Chans | Control |
| System Limits | Minimum: 0, Maximum: 255 |
| Event Source | Yes |
When queried, this parameter returns the keycode value of the last key that was held on the IR remote. As an event, a status message is generated at an interval of approximately 100 ms whenever an key is held on the IR remote.
ir_chan_id Set IR Remote Channel ID
| Channel Type | Virtual Channel |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| Phys Chans | Infrared Remote Input |
| Virt Chans | Control |
| System Limits | Minimum: 0, Maximum: 15 |
| Default | 3 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the channel ID that the specified IR input will respond to.
Polycom HDX Series IR remotes and Polycom RealPresence Group Series IR remotes can be configured to use different channel IDs so that multiple remotes can be used in the same room to control different equipment without interfering with each other. By default, the Polycom IR remote uses channel ID 3. This can be changed by following the instructions in the Polycom HDX or Polycom RealPresence Group Series Administrator's Guide.
ser_baud RS-232 Baud Rate
| Channel Type | Device-Specific System | |||||||||||||||
| Value Type | List | |||||||||||||||
| Read/Write Mode | Read/Write | |||||||||||||||
| Values |
|
|||||||||||||||
| Event Source | No |
This parameter sets the baud rate for the RS-232 port.
Hardware flow control should be enabled for baud rates over
9600 bps (see the ser_flow parameter).
| Command | Response | Description |
|---|---|---|
| set ser_baud 1 9600 |
val ser_baud 1 9600 |
Sets the serial port baud rate on device 1 (the first device) to 9600. |
ser_control_mode Set RS-232 Control Mode
| Channel Type | Device-Specific System | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter sets the mode of operation for the RS-232
port. When set to command, the RS-232 port is
operating as an interface to the SoundStructure command
processor. When set to broadcast the
ser_send parameter can be used to send
arbitrary commands to control other devices connected to the
RS-232 port. In broadcast mode, all received
data is ignored.
| Command | Response | Description |
|---|---|---|
| set ser_control_mode 1 command |
val ser_control_mode 1 command |
Sets the serial port mode on SoundStructure device 1 to command mode. |
| get ser_control_mode 1 |
val ser_control_mode 1 command |
Queries the value of the ser_control_mode on SoundStructure device 1. |
| set ser_control_mode 1 broadcast |
val ser_control_mode 1 broadcast |
Sets the serial port mode on SoundStructure device 1 to broadcast. If you send this command to a SoundStructure device via the serial port and you set the serial port to broadcast mode you will not receive an acknowledgment as the mode of the serial port has been changed. |
ser_flow RS-232 Flow Control
| Channel Type | Device-Specific System | ||||||
| Value Type | List | ||||||
| Read/Write Mode | Read/Write | ||||||
| Values |
|
||||||
| Event Source | No |
This parameter sets the type of flow control that will be used on the RS-232 port. Hardware flow control is recommended for baud rates over 9600 bps.
| Command | Response | Description |
|---|---|---|
| set ser_flow 1 hw |
val ser_flow 1 hw |
Sets the serial port flow control on device 1 (the first device) to 'hw'. |
| set ser_flow 1 none |
val ser_flow 1 none |
Disables the serial port flow control on device 1 (the first device) by setting the flow control to 'none'. |
ser_send Send Arbitrary Data to RS-232 Port
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Write-Only |
| System Limits | Max String Length: 256 |
| Event Source | No |
This parameter is used to broadcast arbitrary commands to
equipment attached to the RS-232 port. If the
ser_control_mode parameter is set to
broadcast for the RS-232 port, then the data in
the string will be sent to the RS-232 port. If it is set to
command then setting this parameter has no
effect.
Special characters may be included in the string by escaping them. Since the string format already uses a backslash to escape double quote characters, a double backslash must be used to escape the special characters for this parameter. The following escape sequences are supported.
| Command | Response | Description |
|---|---|---|
| set ser_send 1 "Hello, World\\r" |
set ser_send 1 "Hello, World\\r" |
Send the string "Hello, World" followed by a carriage return to the RS-232 port on device 1. |
| set ser_send 1 "\\x48\\x65\\x6c\\x6c\\x6f\\x2c\\x20\\x57\\x6f\\x72\\x6c\\x64\\x0d" |
set ser_send 1 "\\x48\\x65\\x6c\\x6c\\x6f\\x2c\\x20\\x57\\x6f\\x72\\x6c\\x64\\x0d" |
Send the same string as the previous example, but using hexadecimal to specify the bytes instead. |
clink_num_attached Number of Devices Attached via Conference Link
| Channel Type | Device-Specific System | |||
| Value Type | Integer | |||
| Read/Write Mode | Read-Only | |||
| Indices |
|
|||
| System Limits | Minimum: 0, Maximum: 32 | |||
| Event Source | No |
This parameter returns the number of devices of a specific type attached to both ConferenceLink ports of the specified device. The index specifies the type of ConferenceLink device to count (1 = all devices, 2 = Polycom table mics, 3 = Polycom ceiling mics, 4 = Polycom video codecs, 5 = Polycom conference phones).
dev_bootloader_ver Bootloader Version
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| System Limits | Max String Length: 24 |
| Event Source | No |
This parameter returns the bootloader version.
dev_firmware_ver Firmware Version
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| System Limits | Max String Length: 24 |
| Event Source | No |
This parameter returns the device's firmware version.
| Command | Response | Description |
|---|---|---|
| get dev_firmware_ver 1 |
val dev_firmware_ver 1 "1.0.0" |
Returns the revision of the firmware for device 1 (the first device). |
dev_hw_eco Hardware ECO Number
| Channel Type | Device-Specific System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 0, Maximum: 255 |
| Event Source | No |
This parameter returns the hardware ECO number for the board. The number does not track the actual ECO number, but rather indicates major ECO changes that we may need to account for in software.
dev_hw_rev Hardware Revision
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| System Limits | Max String Length: 32 |
| Event Source | No |
This parameter returns the hardware revision of the device. Typical values are "A", "B", "C", "D", "E", etc.
dev_led_cycle Cycle Front Panel LED
| Channel Type | Device-Specific System |
| Value Type | Integer |
| Read/Write Mode | Write-Only |
| System Limits | Minimum: 1, Maximum: 60 |
| Event Source | No |
This parameter causes the front panel LED of the specified device to cycle through its colors (yellow-red-green-off) for the specified number of seconds.
dev_ntp_server NTP Server
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read/Write |
| System Limits | Max String Length: 32 |
| Default | pool.ntp.org |
| Event Source | No |
This parameter gets or sets the name of the network time protocol (NTP) server used to set the system time.
dev_plugin_type Plugin Card Type
| Channel Type | Device-Specific System | |||||||||||||||
| Value Type | List | |||||||||||||||
| Read/Write Mode | Read-Only | |||||||||||||||
| Indices |
|
|||||||||||||||
| Values |
|
|||||||||||||||
| Event Source | No |
This parameter returns the type of the device.
dev_status System Status
| Channel Type | Device-Specific System | |||||||||
| Value Type | List | |||||||||
| Read/Write Mode | Read-Only | |||||||||
| Values |
|
|||||||||
| Event Source | No |
This parameter returns the status of the system. A value
of ok indicates that the system is operating
normally. The front-panel LEDs on all the devices will be
green in this condition. A value of warning
indicates that a warning condition has occurred. A warning
condition is usually due to a configuration error that can be
corrected via software. The front-panel LEDs on all of the
devices will be yellow in this condition. A value of
error indicates that an error has occurred that
is most likely due to a hardware failure or some other serious
condition that can't be corrected via software. The front
panel LEDs on one or more of the linked devices will be red in
this condition.
| Command | Response | Description |
|---|---|---|
| get dev_status 1 |
val dev_status "ok" |
Queries the status of the device and in this example, the status was "ok". |
| get dev_status 1 |
val dev_status "warning" |
Queries the status of device 2 which is in a warning state, likely because the configuration project does not match the devices due to hardware changes or unbussing units. |
| get dev_status 4 |
error "invalid device ID" |
Queries the status of a non-existent device which returns an error message. |
dev_temp Internal Temperature
| Channel Type | Device-Specific System | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read-Only | |||
| Indices |
|
|||
| System Limits | Minimum: -40.0, Maximum: 125.0, Resolution: 0.1 | |||
| Event Source | No |
This parameter returns the temperature (in degrees C) from of one of the internal temperature sensors. The temperature sensors have the following locations by index. 1 is at the back right, underneath the plug-in slot. 2 is near the center of the analog input circuitry. 3 is at the front right, in front of the power supply.
dev_temp_status Internal Temperature Status
| Channel Type | Device-Specific System | |||||||||
| Value Type | List | |||||||||
| Read/Write Mode | Read-Only | |||||||||
| Values |
|
|||||||||
| Event Source | Yes |
This parameter returns the temperature status of the
specified device. If the internal temperature sensors
indicate the device is operating within a safe temperature
range, the ok value is returned. The
warning value is returned when the temperature
is in the marginal temperture range. The error
value is returned when the temperature is too high indicating airflow in the equipment rack must be improved.
| Command | Response | Description |
|---|---|---|
| get dev_temp_status 1 |
val dev_temp_status ok |
Queries the temperature status of SoundStructure device 1. |
dev_tp_mode Telepresence Mode
| Channel Type | Device-Specific System |
| Value Type | Boolean |
| Read/Write Mode | Write-Only |
| Event Source | No |
This parameter converts a C16, C12, or C8 to be a telepresence device (T16, T12, or T8) when set to 1. The device will be converted back to a C16, C12, or C8 when set to 0.
Note that when a device is in telepresence mode, setting the sys_factory_reset parameter will cause the device to reboot as part of the factory reset procedure. This is because the DSPs need to re-load the non-telepresence configuration.
dev_type Device Type
| Channel Type | Device-Specific System | |||||||||||||||||||||||||||
| Value Type | List | |||||||||||||||||||||||||||
| Read/Write Mode | Read-Only | |||||||||||||||||||||||||||
| Values |
|
|||||||||||||||||||||||||||
| Event Source | No |
This parameter returns the type of the device.
dev_uptime System Uptime
| Channel Type | Device-Specific System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| System Limits | Max String Length: 16 |
| Event Source | No |
This parameter returns the amount of time since the last reboot. The value returned is formatted as days:hours:minutes:seconds. For example, a value of "247:02:14:31" indicates the system has been running for 247 days, 2 hours, 14 minutes, and 31 seconds.
| Command | Response | Description |
|---|---|---|
| get dev_uptime 1 |
val dev_uptime"62:07:05:36" |
Queries the uptime of the device and this example returns 62 days, 7 hours, 5 minutes, and 36 seconds. |
dev_volt_clink ConferenceLink Supply Voltage
| Channel Type | Device-Specific System |
| Value Type | Floating-Point |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 0.0, Maximum: 50.5, Resolution: 0.1 |
| Event Source | No |
This parameter returns the voltage (in Volts) of the ConferenceLink power supply.
dev_volt_neg_15 -15 V Supply Voltage
| Channel Type | Device-Specific System |
| Value Type | Floating-Point |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.1 |
| Event Source | No |
This parameter returns the voltage (in Volts) of the -15 V power supply.
dev_volt_phantom Phantom Power Supply Voltage
| Channel Type | Device-Specific System | |||
| Value Type | Floating-Point | |||
| Read/Write Mode | Read-Only | |||
| Indices |
|
|||
| System Limits | Minimum: 0.0, Maximum: 50.5, Resolution: 0.1 | |||
| Event Source | No |
This parameter returns the output voltage (in Volts) of the specified phantom power supply. There is one phantom power supply for each bank of four inputs. Thus, there are two banks on an 8x8, three banks on a 12x12, and four banks on a 16x16. The voltage will be 0 for supplies that are not present on a device. Index 1 corresponds to the phantom power supply on mic inputs 1-4, index 2 corresponds to the phantom power supply on mic inputs 5-8, and so on.
dev_volt_pos_15 +15 V Supply Voltage
| Channel Type | Device-Specific System |
| Value Type | Floating-Point |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: -20.0, Maximum: 20.0, Resolution: 0.1 |
| Event Source | No |
This parameter returns the voltage (in Volts) of the +15 V power supply.
sys_bus_id Get ID of connected device
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 1, Maximum: 8 |
| Event Source | No |
This parameter returns the bus ID for the device on which the command is received. This parameter is primarily used by SoundStructure Studio during device discovery so that it can query the ID of the device to which it is connected.
sys_cmd_log_en Enable/disable reporting of cmd/ack in logs
| Channel Type | Global System |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Default | 1 |
| Event Source | No |
This parameter controls whether commands and acknowledgements are reported in the system logs.
| Command | Response | Description |
|---|---|---|
| set sys_cmd_log_en 0 |
val sys_cmd_log_en 0 |
Turns off recording the commands and acknowledgments from being entered into the SoundStructure system logs. |
| set sys_cmd_log_en 1 |
val sys_cmd_log_en 1 |
Enables the recording of SoundStructure commands and acknowledgments to the SoundStructure system logs. |
sys_devices_match Report whether system devices match stored configuration
| Channel Type | Global System |
| Value Type | Boolean |
| Read/Write Mode | Read-Only |
| Event Source | No |
This parameter returns true (1) if the actual
devices in the system match the stored configuration.
sys_factory_reset Restore System to Factory Settings
| Channel Type | Global System |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Event Source | No |
Setting this parameter restores the device to its factory settings, erasing all user data but retaining the current version of firmware.
| Command | Response | Description |
|---|---|---|
| set sys_factory_reset |
val sys_factory_reset |
Returns the SoundStructure system to a factory default state including setting the ethernet mode back to DHCP. If connected over ethernet, you will lose your connection if the IP address changes and you will not see the command acknowledgment. The system does not reboot after a sys_factory_reset. |
sys_last_full_preset Get Last Executed Full Preset
| Channel Type | Global System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Event Source | No |
This parameter returns the name of the last executed full preset.
sys_last_partial_preset Get Last Executed Partial Preset
| Channel Type | Global System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Event Source | No |
This parameter returns the name of the last executed partial preset.
sys_last_preset Get Last Executed Preset
| Channel Type | Global System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| Event Source | No |
This parameter returns the name of the last executed preset (whether partial or full).
sys_meter_period Period of Meter Data Messages
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Read/Write |
| System Limits | Minimum: 1, Maximum: 10000 |
| Default | 1 |
| User Limits Supported | No |
| Event Source | No |
This parameter sets the rate (in milliseconds) at which meter data commands will be generated.
sys_mtrreg_log_en Enable/disable reporting of mtrreg/mtrunreg in logs
| Channel Type | Global System |
| Value Type | Boolean |
| Read/Write Mode | Read/Write |
| Default | 1 |
| Event Source | No |
This parameter controls whether mtrreg and mtrunreg commands and acknowledgements are reported in the system logs.
| Command | Response | Description |
|---|---|---|
| set sys_mtrreg_log_en 0 |
val sys_mtrreg_log_en 0 |
Turns off recording of the meter registration commands from being entered into the SoundStructure system logs. |
| set sys_mtrreg_log_en 1 |
val sys_mtrreg_log_en 1 |
Enables the recording of meter registration commands to the SoundStructure system logs. |
sys_name System Name
| Channel Type | Global System |
| Value Type | String |
| Read/Write Mode | Read/Write |
| System Limits | Max String Length: 256 |
| Default | SoundStructure System |
| Event Source | No |
This parameter sets the name of the system.
sys_num_auth_connections Get number of auth Ethernet connections
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 0, Maximum: 256 |
| Event Source | No |
This parameter returns the total number of Ethernet
connections for which eth_auth_mode is
auth.
sys_num_connections Get number of Ethernet connections
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 0, Maximum: 256 |
| Event Source | No |
This parameter returns the total number of Ethernet connections to the system.
sys_num_devs Number of Devices
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 1, Maximum: 8 |
| Event Source | No |
This parameter returns the total number of devices on the bus.
sys_num_open_connections Get number of open Ethernet connections
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Read-Only |
| System Limits | Minimum: 0, Maximum: 256 |
| Event Source | No |
This parameter returns the total number of Ethernet
connections for which eth_auth_mode is
open.
sys_pause Pause System Execution
| Channel Type | Global System |
| Value Type | Integer |
| Read/Write Mode | Write-Only |
| System Limits | Minimum: 1, Maximum: 5000 |
| Event Source | No |
This parameter pauses system execution for the specified number of milliseconds. Note that the entire command processor is paused, affecting all communication ports.
The typical application for this parameter is inserting pauses between commands in partial preset execution.
sys_plugins_match Report whether system plug-in cards match stored configuration
| Channel Type | Global System |
| Value Type | Boolean |
| Read/Write Mode | Read-Only |
| Event Source | No |
This parameter returns true (1) if the actual
plug-in cards in the system match the stored configuration.
sys_reboot Reset the Device
| Channel Type | Global System |
| Value Type | Void |
| Read/Write Mode | Write-Only |
| Event Source | No |
Setting this parameter causes all linked devices to reboot as if a power-cycle has occurred.
| Command | Response | Description |
|---|---|---|
| set sys_reboot |
val sys_reboot |
Reboots the SoundStructure system. Any settings that were not saved to a preset prior to rebooting will be lost. |
sys_tp_cfg Select a pre-defined telepresence configuration
| Channel Type | Global System | |||||||||||||||
| Value Type | List | |||||||||||||||
| Read/Write Mode | Write-Only | |||||||||||||||
| Values |
|
|||||||||||||||
| Event Source | No |
Setting this parameter causes a pre-defined configuration for telepresence systems to be written over the current configuration.
sys_last_tp_cfg Report the last selected telepresence configuration
| Channel Type | Global System |
| Value Type | String |
| Read/Write Mode | Read-Only |
| System Limits | Max String Length: 256 |
| Event Source | No |
Querying this parameter reports the value of
sys_tp_cfg the last time it was set, along with the
firmware version that was running at the time.