Configuring Tel Profile

The Tel Profiles table lets you configure up to 9Tel Profiles. A Tel Profile is a set of parameters with specific settings which can be assigned to specific calls. The Tel Profiles table includes a wide range of parameters for configuring the Tel Profile. Each of these parameters has a corresponding "global" parameter, which when configured applies to all calls. The main difference, if any, between the Tel Profile parameters and their corresponding global parameters are their default values.

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).

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To configure a Tel Profile:

Open the Tel Profiles table (Setup menu > Signaling & Media tab > Coders & Profiles folder > Tel Profiles).

Click New; the following dialog box appears:

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.

Click Apply.

Tel Profile Table Parameter Descriptions

Parameter

Description

General

'Index'

[Index]

Defines an index number for the new table row.

Note: Each row must be configured with a unique index.

'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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Configure each row with a unique name.

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The parameter value can't contain a forward slash (/).

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The parameter value can't be configured with the character string "any" (upper or lower case).

Signaling

'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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If both the IP Profile and Tel Profile apply to the same call, the coders and common parameters of the Preferred profile are applied to the call.

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If the Preference of the Tel Profile and IP Profile are identical, the Tel Profile parameters are applied.

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If the coder lists of both the IP Profile and Tel Profile apply to the same call, only the coders common to both are used. The order of the coders is determined by the preference.

'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.

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[0] No Fax = (Default) No fax negotiation using SIP signaling. The fax transport method is according to the FaxTransportMode parameter.

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[1] T.38 Relay = Initiates T.38 fax relay.

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[2] G.711 Transport = Initiates fax/modem using the coder G.711 A-law/Mu-law with adaptations (see Note below).

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[3] Fax Fallback = Initiates T.38 fax relay. If the T.38 negotiation fails, the device re-initiates a fax session using the coder G.711 A-law/Mu-law with adaptations (see the Note below).

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[4] G.711 Reject T.38 = Initiates fax/modem using the coder G.711 A-law/Mu-law with adaptations (see Note below), but if the incoming media is of type IMAGE, the device rejects the re-INVITE message for T.38.

Note:

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Fax adaptations (for options 2 and 3):

✔

Echo Canceller = On

✔

Silence Compression = Off

✔

Echo Canceller Non-Linear Processor Mode = Off

✔

Dynamic Jitter Buffer Minimum Delay = 40

✔

Dynamic Jitter Buffer Optimization Factor = 13

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If the device initiates a fax session using G.711 (option 2 or 3), a 'gpmd' attribute is added to the SDP in the following format:

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For A-law: 'a=gpmd:8 vbd=yes;ecan=on'

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For Mu-law: 'a=gpmd:0 vbd=yes;ecan=on'

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When the parameter is set to 1, 2, or 3, the parameter FaxTransportMode is ignored.

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When the parameter is set to 0, T.38 might still be used without the control protocol's involvement. To completely disable T.38, set FaxTransportMode to a value other than 1.

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For more information on fax transport methods, see Fax/Modem Transport Modes.

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The corresponding global parameter is [IsFaxUsed].

'Enable Digit Delivery'

digit-delivery

[EnableDigitDelivery]

Enables the Digit Delivery feature, which sends DTMF digits of the called number to the phone line (device'sdigital B-channel) after the call is answered for IP-to-Tel calls.

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[0] Disable (default)

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[1] Enable

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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The corresponding global parameter is [EnableDigitDelivery].

'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].

'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.

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[0] Disable (default)

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[1] Enable = When enabled, the feature operates as follows:

A Tel-to-IP call is established (connected) by the device for a B-channel.

The device receives a SIP BYE (i.e., IP side ends the call) and plays a reorder tone to the PSTN side for the duration configured by the CutThroughTimeForReOrderTone parameter. The device releases the call towards the IP side (sends a SIP 200 OK).

The PSTN side, for whatever reason, remains off-hook.

If a new IP call is received for this B-channel after the reorder tone has ended, the device “cuts through” the channel and connects the call immediately (despite the B-channel being in physical off-hook state) without playing a ring tone. If an IP call is received while the reorder tone is played, the device rejects the call.

Note:

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If the parameter is disabled and the PSTN side remains in off-hook state after the IP call ends the call, the device releases the call after 60 seconds.

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A special CAS table can be used to report call status events (Active/Idle) to the PSTN side during Cut Through mode (see Configuring CAS State Machines).

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The corresponding global parameter is [DigitalCutThrough].

'Call Priority Mode'

call-priority-mode

[CallPriorityMode]

Defines call priority handling.

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[0] Disable (default).

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[1] MLPP = Enables MLPP Priority Call handling. MLPP prioritizes call handling whereby the relative importance of various kinds of communications is strictly defined, allowing higher precedence communication at the expense of lower precedence communications. Higher priority calls override less priority calls when, for example, congestion occurs in a network.

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[2] Emergency = Enables Preemption of IP-to-Tel E911 emergency calls. If the device receives an E911 call and there are unavailable channels to receive the call, the device terminates one of the channel calls and sends the E911 call to that channel. The preemption is done only on a channel belonging to the same Trunk Group for which the E911 call was initially destined and if the channel select mode (configured by the ChannelSelectMode parameter) is set to other than By Dest Phone Number (0). The preemption is done only if the incoming IP-to-Tel call is identified as an emergency call. The device identifies emergency calls by one of the following:

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The destination number of the IP call matches one of the numbers configured by the EmergencyNumbers parameter. (For E911, you must configure the parameter to "911".)

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The incoming SIP INVITE message contains the “emergency” value in the Priority header.

Note:

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For more information, see Pre-empting Existing Call for E911 IP-to-Tel Call.

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The corresponding global parameter is [CallPriorityMode].

Behavior

'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.

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[0] Disable

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[1] Enable (default)

Note:

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The parameter is applicable only to CAS.

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The corresponding global parameter is [DisconnectOnBusyTone].

'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 254. The default is 10 seconds for digital interfaces. Note that the Web interface denotes the default value as "255".

Note:

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The selected busy or reorder tone is according to the SIP release cause code received from IP.

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The parameter is also applicable to CAS.

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The parameter is applicable to ISDN when the [PlayBusyTone2ISDN] parameter is set to 2.

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The corresponding global parameter is [TimeForReorderTone].

'Enable Voice Mail Delay'

enable-voice-mail-delay

[EnableVoiceMailDelay]

Enables and disables voice mail services.

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[0] Disable

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[1] Enable (default)

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.

'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.

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[0] Disable (default)

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[1] Enable

Note: The corresponding global parameter is [SwapTEl2IPCalled&CallingNumbers].

Voice

'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].

'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].

'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].

'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.

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[0] Disable (default)

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[1] Enable

Note:

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For more information on AGC, see Automatic Gain Control (AGC).

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The corresponding global parameter is [EnableAGC].

IP Settings

'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.

'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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For more information on DiffServ, see Configuring Class-of-Service QoS.

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The corresponding global parameter is [PremiumServiceClassMediaDiffServ].

'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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For more information on DiffServ, see Configuring Class-of-Service QoS.

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The corresponding global parameter is [PremiumServiceClassControlDiffServ].

'Enable Early Media'

early-media

[EnableEarlyMedia]

Enables the Early Media feature, which sends media (e.g., ringing) before the call is established.

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[0] Disable (default)

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[1] Enable

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The device sends a SIP 18x response with SDP, allowing the media stream to be established before the call is answered.

Note:

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The inclusion of the SDP in the 18x response depends on the ISDN Progress Indicator (PI). The SDP is sent only if PI is set to 1 or 8 in the received Proceeding, Alerting, or Progress messages. See also the ProgressIndicator2IP parameter, which if set to 1 or 8, the device behaves as if it received the ISDN messages with the PI.

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CAS: See the ProgressIndicator2IP parameter.

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ISDN: Sending a 183 response depends on the ISDN PI. It is sent only if PI is set to 1 or 8 in the received Proceeding or Alerting messages. Sending 183 response also depends on the ReleaseIP2ISDNCallOnProgressWithCause parameter, which must be set to any value other than 2.

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See also the IgnoreAlertAfterEarlyMedia parameter. The parameter allows, for example, to interwork Alert with PI to SIP 183 with SDP instead of 180 with SDP.

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You can also configure early SIP 183 response immediately upon the receipt of an INVITE, using the EnableEarly183 parameter.

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The corresponding global parameter is [EnableEarlyMedia].

'Progress Indicator to IP'

prog-ind-to-ip

[ProgressIndicator2IP]

Defines the progress indicator (PI) sent to the IP.

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[-1] = (Default) Not configured:

✔

The PI received in ISDN Proceeding, Progress, and Alerting messages is used, as described in the options below.

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[0] No PI =

✔

For IP-to-Tel calls, the device sends 180 Ringing response to the IP after receiving an ISDN Alerting, or for CAS after placing a call to the PBX/PSTN.

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[1] PI = 1 =

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For IP-to-Tel calls, if the parameter EnableEarlyMedia is set to 1, the device sends 180 Ringing with SDP in response to an ISDN Alerting or it sends a 183 Session Progress message with SDP in response to only the first received ISDN Proceeding or Progress message after a call is placed to PBX/PSTN over the trunk.

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[8] PI = 8 = Same as PI = 1.

Note: The corresponding global parameter is [ProgressIndicator2IP].

Echo Canceler

'Echo Canceler'

echo-canceller

[EnableEC]

Enables the device's Echo Cancellation feature (i.e., echo from voice calls is removed).

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[0] Disable

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[1] Line Echo Canceller (default)

For more information on echo cancellation, see Configuring Echo Cancellation.

Note: The corresponding global parameter is [EnableEchoCanceller].

'EC NLP Mode'

echo-canceller-nlp-mode

[ECNlpMode]

Enables Non-Linear Processing (NLP) mode for echo cancellation.

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[0] Adaptive NLP = (Default) NLP adapts according to echo changes

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[1] Disable NLP

Note: The corresponding global parameter is [ECNLPMode].

Jitter Buffer

'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].

'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.

'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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For data (fax and modem) calls, configure the parameter to 12.

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The corresponding global parameter is [DJBufOptFactor].