Configuring Tel Profile
The Tel Profiles table lets you configure up to
Tel Profiles provide high-level adaptation when the device interworks between different equipment and protocols (at both the Tel and IP sides), each of which may require different handling by the device. For example, if specific channels require the use of the G.711 coder, you can configure a Tel Profile with this coder and assign it to these channels.
To use your Tel Profile for specific calls, you need to assign it to specific channels (trunks or endpoints) in the Trunk Groups table (see Configuring Trunk Groups).
The following procedure describes how to configure Tel Profiles through the Web interface. You can also configure it through ini file [TelProfile] or CLI (configure voip > coders-and-profiles tel-profile).
➢ | To configure a Tel Profile: |
1. | Open the Tel Profiles table (Setup menu > Signaling & Media tab > Coders & Profiles folder > Tel Profiles). |
2. | Click New; the following dialog box appears: |
3. | Configure a Tel Profile according to the parameters described in the table below. For a description of each parameter, refer to the corresponding "global" parameter. |
4. | Click Apply. |
Tel Profile Table Parameter Descriptions
Parameter |
Description |
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'Index' [Index] |
Defines an index number for the new table row. Note: Each row must be configured with a unique index. |
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'Name' profile-name [ProfileName] |
Defines a descriptive name, which is used when associating the row in other tables. The valid value is a string of up to 40 characters. Note:
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Signaling |
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'Profile Preference' tel-preference [TelPreference] |
Defines the priority of the Tel Profile, where 1 is the lowest priority and 20 the highest priority. Note:
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'Fax Signaling Method' fax-sig-method [IsFaxUsed] |
Defines the SIP signaling method for establishing and transmitting a fax session when the device detects a fax.
Note:
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'Enable Digit Delivery' digit-delivery [EnableDigitDelivery] |
Enables the Digit Delivery feature, which sends DTMF digits of the called number to the phone line (device's
Digital interfaces: If the called number in IP-to-Tel call includes the characters 'w' or 'p', the device places a call with the first part of the called number (before 'w' or 'p') and plays DTMF digits after the call is answered. If the character 'w' is used, the device waits for detection of a dial tone before it starts playing DTMF digits. For example, if the called number is '1007766p100', the device places a call with 1007766 as the destination number, then after the call is answered it waits 1.5 seconds ('p') and plays the rest of the number (100) as DTMF digits. Additional examples: 1664wpp102, 66644ppp503, and 7774w100pp200. Note:
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'Dial Plan Index' dial-plan-index [DialPlanIndex] |
Defines the Dial Plan index to use in the external Dial Plan file. Note: The corresponding global parameter is [DialPlanIndex]. |
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'Digital Cut Through' digital-cut-through [DigitalCutThrough] |
Enables PSTN CAS channels/endpoints to receive incoming IP calls even if the B-channels are in off-hook state.
Note:
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'Call Priority Mode' call-priority-mode [CallPriorityMode] |
Defines call priority handling.
Note:
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Behavior |
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'Disconnect Call on Detection of Busy Tone' disconnect-on-busy-tone [DisconnectOnBusyTone] |
Enables the device to disconnect the call upon detection of a busy or reorder (fast busy) tone.
Note:
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'Time For Reorder Tone' time-for-reorder-tone [TimeForReorderTone] |
Defines the duration (in seconds) that the device plays a busy or reorder tone before releasing the line.
The valid range is 0 to Note:
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'Enable Voice Mail Delay' enable-voice-mail-delay [EnableVoiceMailDelay] |
Enables and disables voice mail services.
The parameter is useful if you want to disable voice mail services per Trunk Group to eliminate the phenomenon of call delay on Trunks that do not implement voice mail when voice mail is configured using the global parameter, VoiceMailInterface. |
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'Swap Tel To IP Phone Numbers' swap-teltoip-phone-numbers [SwapTelToIpPhoneNumbers] |
Enables the device to swap the calling and called numbers received from the Tel side (for Tel-to-IP calls). The SIP INVITE message contains the swapped numbers.
Note: The corresponding global parameter is [SwapTEl2IPCalled&CallingNumbers]. |
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'IP-to-Tel Cut-Through Call Mode' ip2tel-cutthrough_call_behavior [IP2TelCutThroughCallBehavior] |
Enables the Cut-Through feature, which allows phones connected to the device’s FXS ports to automatically receive IP calls (if there is no other currently active call) even when in off-hook state (and no call is currently active).
Note:
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Voice |
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'DTMF Volume' dtmf-volume [DtmfVolume] |
Defines the DTMF gain control value (in decibels) to the Tel side. The valid range is -31 to 0 dB. The default is -11 dB. Note: The corresponding global parameter is [DTMFVolume]. |
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'Input Gain' input-gain [InputGain] |
Defines the pulse-code modulation (PCM) input (received) gain control level (in decibels), which is the level of the received signal for Tel-to-IP calls. The valid range is -32 to 31 dB. The default is 0 dB. Note: The corresponding global parameter is [InputGain]. |
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'Voice Volume' voice-volume [VoiceVolume] |
Defines the voice gain control (in decibels), which is the level of the transmitted signal for IP-to-Tel calls. The valid range is -32 to 31 dB. The default is 0 dB. Note: The corresponding global parameter is [VoiceVolume]. |
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'Enable AGC' enable-agc [EnableAGC] |
Enables the Automatic Gain Control (AGC) feature. The AGC feature automatically adjusts the level of the received signal to maintain a steady (configurable) volume level.
Note:
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Analog |
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'Enable Polarity Reversal' polarity-rvrsl [EnableReversePolarity] |
Enables the Polarity Reversal feature for call release.
Note:
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'Enable Current Disconnect' current-disconnect [EnableCurrentDisconnect] |
Enables call release upon detection of a Current Disconnect signal.
Note:
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'DID Wink' enable-did-wink [EnableDIDWink] |
Enables Direct Inward Dialing (DID) using Wink-Start signaling, typically used for signaling between an E-911 switch and the PSAP.
For example: (Wink) KP I(I) xxx-xxxx ST (Off Hook) Where:
Note: The FXO interface generates such MF digits when the Enable911PSAP parameter is set to 1.
Note:
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'Enable 911 PSAP' enable-911-psap [Enable911PSAP] |
Enables the support for the E911 DID protocol, according to the Bellcore GR-350-CORE standard. The protocol defines signaling between E911 Tandem Switches and the PSAP, using analog loop-start lines. The device's FXO interface can be used instead of an E911 switch, connected directly to PSAP DID loop-start lines.
Note:
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IP Settings |
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'Coders Group' coders-group [CodersGroupName] |
Assigns a Coder Group, which defines audio (voice) coders that can be used for the endpoints associated with the Tel Profile. To configure Coders Groups, see Configuring Coder Groups. |
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'RTP IP DiffServ' rtp-ip-diffserv [IPDiffServ] |
Defines the DiffServ value for Premium Media class of service (CoS) content. The valid range is 0 to 63. The default is 46. Note:
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'Signaling DiffServ' signaling-diffserv [SigIPDiffServ] |
Defines the DiffServ value for Premium Control CoS content (Call Control applications). The valid range is 0 to 63. The default is 40. Note:
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'Enable Early Media' early-media [EnableEarlyMedia] |
Enables the Early Media feature, which sends media (e.g., ringing) before the call is established.
Note:
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'Progress Indicator to IP' prog-ind-to-ip [ProgressIndicator2IP] |
Defines the progress indicator (PI) sent to the IP.
Note: The corresponding global parameter is [ProgressIndicator2IP]. |
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Echo Canceler |
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'Echo Canceler' echo-canceller [EnableEC] |
Enables the device's Echo Cancellation feature (i.e., echo from voice calls is removed).
For more information on echo cancellation, see Configuring Echo Cancellation. Note: The corresponding global parameter is [EnableEchoCanceller]. |
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'EC NLP Mode' echo-canceller-nlp-mode [ECNlpMode] |
Enables Non-Linear Processing (NLP) mode for echo cancellation.
Note: The corresponding global parameter is [ECNLPMode]. |
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Jitter Buffer |
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'Dynamic Jitter Buffer Minimum Delay' jitter-buffer-minimum-delay [JitterBufMinDelay] |
Defines the minimum delay (in msec) of the device's dynamic Jitter Buffer. The valid range is 0 to 150. The default delay is 10. For more information on Jitter Buffer, see Configuring the Dynamic Jitter Buffer. Note: The corresponding global parameter is [DJBufMinDelay]. |
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'Dynamic Jitter Buffer Maximum Delay' jitter-buffer-maximum-delay [JitterBufMaxDelay] |
Defines the maximum delay (in msec) for the device's Dynamic Jitter Buffer. The default is 300. |
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'Dynamic Jitter Buffer Optimization Factor' jitter-buffer-optimization-factor [JitterBufOptFactor] |
Defines the Dynamic Jitter Buffer frame error/delay optimization factor. The valid range is 0 to 12. The default factor is 10. For more information on Jitter Buffer, see Configuring the Dynamic Jitter Buffer. Note:
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Analog MWI |
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'MWI Analog Lamp' mwi-analog-lamp [MWIAnalog] |
Enables the visual display of message waiting indications (MWI).
Note:
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'MWI Display' mwi-display [MWIDisplay] |
Enables sending MWI information to the phone display.
Note:
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'MWI Notification Timeout' mwi-ntf-timeout [MWINotificationTimeout] |
Defines the maximum duration (timeout) that a message MWI is displayed on endpoint equipment (phones LED, screen notification or voice tone). When the timeout expires, the MWI is removed. However, each time a new MWI is sent to the endpoint, the timeout restarts its countdown again. For example, assume the timeout is configured to 10 seconds and the timeout has 2 seconds left until the current MWI is removed. If the endpoint now receives a new MWI, the timeout starts counting once again from 10 seconds, displaying both MWIs until the timeout expires. The valid value range is 0 to 2,000,000 seconds, where 0 means unlimited display (default). Note:
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Analog FXO |
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'Two Stage Dial' is-two-stage-dial [IsTwoStageDial] |
Defines the dialing mode for IP-to-Tel (FXO) calls.
Note:
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'FXO Double Answer' fxo-double-answer [EnableFXODoubleAnswer] |
Enables the FXO Double Answer feature, which rejects (disconnects) incoming (FXO) Tel-to-IP collect calls and signals (informs) this call denial to the PSTN.
Note:
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'FXO Ring Timeout' fxo-ring-timeout [FXORingTimeout] |
Defines the delay (in msec) before the device generates a SIP INVITE (call) to the IP side upon detection of a RING_START event from the Tel (FXO) side. This occurs instead of waiting for a RING_END event. The feature is useful for telephony services that employ constant ringing (i.e., when no RING_END is sent). For example, Ringdown circuit is a service that sends a constant ringing current over the line, instead of cadence-based 2 seconds on, 4 seconds off. For example, when a telephone goes off-hook, a phone at the other end instantly rings. If a RING_END event is received before the timeout expires, the device doesn't initiate a call and ignores the detected ring. The device ignores RING_END events detected after the timeout expires. The valid value range is 0 to 50 (msec), in steps of 100-msec. For example, a value of 50 represents 5 sec. The default value is 0 (i.e., standard ring operation - the FXO interface sends an INVITE upon receipt of the RING_END event). Note:
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'Flash Hook Period' flash-hook-period [FlashHookPeriod] |
Defines the hook-flash period (in msec) for Tel and IP sides. For the IP side, it defines the hook-flash period reported to the IP. For the analog side, it defines the following:
The valid range is 25 to 3,000. The default is 700. Note:
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'FXO Consultative Call Transfer' fxo-consult-call-transfer [FXOConsultCallTransfer] |
Enables support for consultative call transfers (initiated by PSAP operators) for emergency (NG9-1-1) calls, based on the NENA i3 Standard for Next Generation 9‑1‑1 (NENA-STA-010.2-2016). When the device receives a SIP INVITE request that is a 911 call, it sends it to the PSAP operator through the FXO port interface. The PSAP operator then puts the 911 caller on hold and establishes a new call with another party (e.g., emergency provider). The PSAP operator then transfers the 911 caller to the new call party. The device uses a SIP REFER message to bridge the 911 caller with the new call party when the PSAP operator goes on hook.
Note:
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