When you configure voice ports for a voice policy, you configure options that define how the system augments and manipulates calls for the voice port endpoint type.

You can configure the following call functionality policy options, depending on the type of voice card you are using:

• Trunk Group— Use these options to configure voice ports as a member of a trunk group for the card. You can configure one trunk group for voice card. The following table describes these options.

  Table 12. Trunk Group Options for Voice Ports

  Option	Description	Cisco IOS CLI Equivalent
  Add New Trunk Group	Click to add a trunk group for the selected card. You can add one trunk group for a voice port.	—
  Copy from Existing	Click to copy an existing trunk group to a new trunk group. In the box that appears, change the name if desired, select a trunk group, and click Copy.	—
  Name	Name of the trunk group. The name can contain up to 32 characters.	trunk group name
  Hunt-Scheme	Select the hunt scheme in the trunk group for outgoing calls: least-idle both—Searches for an idle channel with the shortest idle time least-idle even—Searches for an idle even-numbered channel with the shortest idle time least-idle odd—Searches for an idle odd-numbered channel with the shortest idle time least-used both—Searches for a trunk group member that has the highest number of available channels (applies only to PRI ISDN cards) least-used even—Searches for a trunk group member that has the highest number of available even-numbered channels (applies only to PRI ISDN cards) least-used odd—Searches for a trunk group member that has the highest number of available odd-numbered channels (applies only to PRI ISDN cards) longest-idle both—Searches for an idle odd-numbered channel with the longest idle time longest-idle even—Searches for an idle channel that has the highest number of available even-numbered channels longest-idle odd—Searches for an idle channel that has the highest number of available odd-numbered channels round-robin both—Searches trunk group members in turn for an idle channel, starting with the trunk group member that follows the last used member round-robin even—Searches trunk group member in turn for an idle even-numbered channel, starting with the trunk group member that follows the last used member round-robin odd—Searches trunk group member in turn for an idle odd-numbered channel, starting with the trunk group member that follows the last used member sequential-both—Searches for an idle channel, starting with the trunk group member with the highest preference within the trunk group sequential-even—Searches for an idle even-numbered channel, starting with the trunk group member with the highest preference within the trunk group sequential-odd—Searches for an idle odd-numbered channel, starting with the trunk group member with the highest preference within the trunk group random—Searches for a trunk group member at random and selects a channel from the member at random Default: least-used both	trunk group name hunt-scheme least-idle [both | even | odd ] hunt-scheme least-used [both | even | odd ] hunt-scheme longest-idle [both | even | odd ] hunt-scheme round-robin [both | even | odd ] hunt-scheme sequential [both | even | odd ] hunt-scheme random
  Max Calls	Enter the maximum number of calls that are allowed for the trunk group. If you do not enter a value, there is no limit on the number of calls. If the maximum number of calls is reached, the trunk group becomes unavailable for more calls. In field—Enter the maximum number of incoming calls that are allowed for this trunk group Out field— Enter the maximum number of outgoing calls that are allowed for this trunk group Valid range for both fields: integers 0 through 1000.	trunk group name max-calls voice number-of-calls direction [ in | out]
  Max-Retry	Select the maximum number of outgoing call attempts that the trunk group makes if an outgoing call fails. If you do not enter a value and a call fails, the system does not attempt to make the call again. Valid range: integers 1 through 5.	trunk group name max-retry attempts
  Save Trunk Group	Click to save the Trunk Group that you configured.	—

• Translation Profile—Use these options to configure translation rules for calling and called numbers. The following table describes these options.

  Table 13. Translation Profile Options for Calling and Called Numbers

  Option	Description	Cisco IOS CLI Equivalent
  Add New Translation Profile	Click to add a translation profile for the selected card. You can create up to two translation profiles for this endpoint.	voice translation-profile name
  Copy from Existing	Click to copy an existing translation profile to a new translation profile. In the box that appears, change the name if desired, select a called translation rule and a calling translation rule, and click Copy.	—
  Calling	Click to configure translation rules for the number that is calling in. The Translation Rules pane displays.	translate calling translation-rule-number
  Called	Click to configure translation rules for the number that is being called. The Translation Rules pane displays.	translate called translation-rule-number
  Translation Rules pane	Click Add New to create a translation rule. Alternatively, you can click Copy From Existing to copy an existing translation rule to a new translation rule. In the box that appears, change the name if desired, select a called translation rule and a calling translation rule, and click Copy. In the Translation Rule Number field, enter a unique number that designates the precedence for this rule. Valid range: integers 1 through 100. (Optional) To copy existing translation rules from a CSV file, click Import. Continue to add rules or click Finish. For detailed information about this file, see . Click Add Rule. In the Match field, enter the string that you want the translation rule to affect. Enter the string in regular expression format beginning and ending with a slash (/). For example, /^9/. To include the backslash character (\) in a match string, precede the backslash with a backslash. From the Action drop-down list, select the action that the system performs for calls that match the string in the Match field. The Reject option causes the system to reject the call. The Replace option causes the system to replace the match number with a value that you specify. If you select the Replace action, in the Replace field that displays, enter the string to which to translate the matched string. Enter the number in regular expression format beginning and ending with a slash (/). For example, //, which indicates a replacement of no string. To include the backslash character (\) in a replace string, precede the backslash with a backslash. As an example, if you specify a match string of /^9/ and a replace string of //, the system removes the leading 9 from calls with a number that begins with 9. In this case, the system translates 914085551212 to 14085551212. Click Save. Add more translation rules as needed. (Optional) Click Export to save the translation rules that you created in a CSV file. Click Finish at the bottom of the pane.	voice translation-rule number Match and Replace Rule: rule precedence /match-pattern/ / replace-pattern/ Reject Rule: rule precedence reject /match-pattern/

• Station ID—Use these options to configure the name and number for caller ID display. The following table describes these options.

  Table 14. Station ID Options

  Option	Description	Cisco IOS CLI Equivalent
  Station Name	Enter the name of the station. The station name can contain up to 50 letters, numbers, and spaces, dashes (-), and underscores (_).	station-id name name
  Station Number	Enter the phone number of the station in E.164 format. The station number can contain up to 15 numeric characters.	station-id number number

• Line Params—Use these options to configure line parameters on the card for voice quality. The following table describes these options.

  Table 15. Line Params Options

  Option	Description	Cisco IOS CLI Equivalent
  Gain	Enter the gain, in dB, for voice input. Valid range: –6 through 14. Default: 0	input gain decibels
  Attenuation	Enter the amount of attenuation, in dB, for transmitted voice output. Valid range: –6 through 14. Default: 3.	output attenuation decibels
  Echo Canceller	Select Enable to apply echo cancellation to voice traffic. By default, this option is enabled.	echo-cancel enable
  Voice Activity Detection (VAD)	Select Enable to apply VAD to voice traffic. By default, this option is enabled.	vad
  Compand Type	Select the companding standard to be used to convert between analog and digital signals in PCM systems (U-law or A-law). Default: U-Law.	compand-type {u-law | a-law}
  Impedance	This field does not apply to PRI ISDN cards. Select the terminating impedance for calls. Default: 600r.	impedance {600c | 600r 900c | 900r | complex1 | complex2 | complex3 | complex4 | complex5 | complex6}
  Call Progress Tone	Select the locale for call progress tones.	cptone locale

• Tuning Params—Use these options to configure parameters for signaling between voice ports and another instrument. The following table describes these options.

  Table 16. Tuning Params Options

  Option	Description	Cisco IOS CLI Equivalent
  Tuning Params Options for FXO Cards
  Pre Dial Delay	Enter the delay, in seconds, of the delay on the FXO interface between the beginning of the off-hook state and the initiation of DTMF signaling. Valid range: 0 through 10. Default: 1.	pre-dial-delay seconds
  Supervisory Disconnect	Select the type of tone that indicates that a call has been released and that a connection should be disconnected: Anytone—Any tone indicates a supervisory disconnect Signal—A disconnect signal indicates a supervisory disconnect Dualtone—A dual-tone indicates a supervisory disconnect Default: Signal.	Anytone: supervisory disconnect anytone Signal: supervisory disconnect Dualtone: supervisory disconnect dualtone {mid-call | pre-connect}
  Dial Type	Select the dialing method for outgoing calls: pulse—Pulse dialer dtmf—Dual-tone multifrequency dialer mf—Multifrequency dialer Default: dtmf.	dial-type {dtmf | pulse | mf}
  Timing Sup-Disconnect	Enter the minimum time, in milliseconds, that is required to ensure that an on-hook indication is intentional and not an electrical transient on the line before a supervisory disconnect occurs (based on power denial signaled by the PSTN or PBX). Valid range: 50 through 1500. Default: 350.	timing sup-disconnect milliseconds
  Battery Reversal	Battery reversal reverses the battery polarity on a PBX when a call connects, then changes the battery polarity back to normal when the far-end disconnects. Select Answer to configure the port to support answer supervision by detection of battery reversal. Select Detection Delay to configure the delay time after which the card acknowledges a battery-reversal signal, then enter the delay time in milliseconds. Valid range: 0 through 800. Default: 0 (no delay). If an FXO port or its peer FXS port does not support battery reversal, do not configure battery reversal options to avoid unpredictable behavior.	battery-reversal [answer] battery-reversal-detection-delay milliseconds
  Timing Hookflash out	Enter the duration, in milliseconds, of hookflash indications that the gateway generates on the FXO interface. Valid range: 50 through 1550. Default: 400.	timing hookflash-out milliseconds
  Timing Guard out	Enter the number of milliseconds after a call disconnects before another outgoing call is allowed. Valid range: 300 through 3000. Default: 2000.	timing guard-out milliseconds
  Tuning Params Options for FXS Cards
  Timing Hookflash In	Enter the minimum and maximum duration, in milliseconds, of an on-hook condition to be interpreted as a hookflash by the FXS card. Valid range for minimum duration: 0 through 400. Default minimum value: 50. Valid range for maximum duration: 50 through 1500. Default maximum value: 1000.	timing hookflash-in maximum-milliseconds minimum-milliseconds
  Pulse Digit Detection	To enable pulse digit detection at the beginning of a call, select Yes. Default: Yes.	pulse-digit-detection
  Loop Length	Select the length for signaling on FXS ports (Long or Short). Default: Short.	loop-length [long | short]
  Ring	Frequency—Select the frequency, in Hz, of the alternating current that, when applied, rings a connected device. Default: 25. DC Offset—Applies only if Loop Length is set to Long. Select the voltage threshold below which a ring does not sound on devices. Valid values: 10-volts, 20-volts, 24-volts, 30-volts, and 35-volts.	ring frequency number ring dc-offset number
  Ringer Equivalence Number (REN)	Select the REN for calls that this card processes. This number specifies the loading effect of a telephone ringer on a line. Valid range: 1 through 5. Default: 1.	ren number

• Supervisory Disconnect—Use these options to configure parameters for supervisory disconnect events. The following table describes these options.

  Table 17. Supervisory Disconnect Options

  Option	Description	Cisco IOS CLI Equivalent
  Add New Supervisory Disconnect	Click to add a supervisory disconnect event.	—
  Mode	Choose the mode for the supervisory disconnect event: Custom CPTone—Provides options for configuring cptone detection parameters for a supervisory disconnect event Dual Tone Detection Params— Provides options for configuring dual-tone detection parameters for a supervisory disconnect event	voice class custom-cptone cptone-name voice class dualtone-detect-params tag
  Supervisory Name	Applies to Custom CPTone mode. Enter a name for the supervisory disconnect event. The name can contain up to 32 characters. Valid characters are letters, numbers, dashes (-), and underscores (_).	voice class custom-cptone cptone-name
  Dualtone	Applies to Custom CPTone mode. Select the type of dual-tone that causes a disconnect. Options are: Busy Disconnect Number Unobtainable Out of Service Reorder Ringback	dualtone {ringback |busy | reorder | out-of-service | number-unobtainable | disconnect}
  Cadence	Applies to Custom CPTone mode. Enter the cadence interval, in milliseconds, of the dual-tones that cause a disconnect. Enter the cadence as an on/off value pair, separated with a space. You can enter up to 4 on/off value pairs, separated with a space.	cadence cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time [cycle-3-on-time cycle-3-off-time [ cycle-4-on-time cycle-4-off-time ]]]
  Dualtone Frequency	Applies to Custom CPTone mode. Enter the frequency, in Hz, of each tone in the dual-tone. Valid range for each tone is 300 through 3600.	frequency frequency-1 [frequency-2]
  Supervisory Number	Applies to Custom Dual Tone Detection Params mode. Enter a unique number to identify dual-tone detection parameters. Valid range: 1 through 10000.	voice class dualtone-detect-params tag-number
  Cadence-Variation	Applies to Custom Dual Tone Detection Params mode. Enter the maximum time, in milliseconds, by which the tone onset can vary from the onset time and still be detected. The system multiplies the value that you enter by 10. Valid range: 0 through 200 in units of 10. Default: 10.	cadence-variation time
  Frequency	Applies to Custom Dual Tone Detection Params mode. Max Delay—Enter the maximum delay, in milliseconds, before a supervisory disconnect is performed after the dual-tone is detected. The system multiplies the value that you enter by 10. Valid range: 0 through 100 in units of 10. Default: 10. Max Deviation—Enter the maximum deviation, in Hz, by which each tone can deviate from configured frequencies and be detected. Valid range: 10 through 125. Default: 10. Max Power—Enter the power of the dual-tone, in dBm0, above which a supervisory disconnect is no detected. Valid range: 0 through 20. Default: 10. Min Power— Enter the power of the dual-tone, in dBm0, below which a supervisory disconnect is not detected. Valid range: 10 through 35. Default: 30. Power Twist—Enter difference, in dBm0, between the minimum power and the maximum power of the dual-tone above which a supervisory disconnect is not detected. Valid range: 0 through 15. Default: 6.	freq-max-delay time freq-max-deviation hertz freq-max-power dBm0 freq-min-power dBm0 freq-power-twist dBm0
  Save	Click to save the supervisory disconnect information that you configured.	—

• DID Timers—Use these options to configure timers for DID calls. The following table describes these options.

  Table 18. DID Timers Options

  Option	Description	Cisco IOS CLI Equivalent
  Wait Before Wink	Enter the amount of time, in milliseconds, that the card waits after receiving a call before sending a wink signal to notify the remote side that it can send DNIS information. Valid range: 100 through 6500. Default: 550.	timing wait-wink milliseconds
  Wink Duration	Enter the maximum amount of time, in milliseconds, of the wink signal for the card. Valid range: 50 through 3000. Default: 200.	timing wait-duration milliseconds
  Clear Wait	Enter the minimum amount of time, in milliseconds, between an inactive seizure signal and the call being cleared for the card. Valid range: 200 through 2000. Default: 400.	timing clear-wait milliseconds
  Dial Pulse Min Delay	Enter the amount of time, in milliseconds, between wink-like pulses for the card. Valid range: 0 or 140 through 5000. Default: 140.	timing dial-pulse min-delay milliseconds
  Answer Winkwidth	Enter the minimum delay time, in milliseconds, between the start of an incoming seizure and the wink signal. Valid range: 110 through 290. Default: 210.	timing answer-winkwidth milliseconds

To configure voice ports for a voice policy, follow these steps:

1. From the Cisco SD-WAN Manager menu, choose Configuration > Unified Communications.

2. Click Add Voice Policy, and choose Voice Ports in the left pane.

3. From the Add Voice Ports Policy Profile drop-down list, select Create New.

   Alternatively, you can select Copy from Existing to copy an existing voice policy to a new voice policy. In the box that appears, select the name of the policy profile to copy, enter a new name for the profile if desired, and click Copy.

4. Select FXO, FXS, PRI ISDN, or FXS DID to specify the type of voice port that the policy is for.

5. Select the types of call functionality policy options that you want to configure from the list of options that displays, and click Next. These option types include the following:

   • Trunk Group—Available for FXO, FXS, FXS DID, and PRI ISDN cards.

     Use these options to configure voice ports as a member of a trunk group for the card.

   • Translation Profile—Available for FXO, FXS, PRI ISDN, and FXS DID cards.

     Use these options to configure translation rules for calling and called numbers.

   • Station ID—Available for FXO, FXS, and FXS DID cards.

     Use these options to configure the name and number for caller ID display.

   • Line Params—Available for FXO, FXS, PRI ISDN, and FXS DID cards.

     Use these options to configure line parameters on the card for voice quality.

   • Tuning Params—Available for FXO and FXS cards.

     Use these options to configure parameters for signaling between voice ports and another instrument.

   • Supervisory Disconnect—Available for FXO cards.

     Use these options to configure parameters for supervisory disconnect events. These events provide an indication that a call has disconnected.

   • DID Timers—Available for FXS DID cards.

     Use these options to configure timers for DID calls.

6. In the page that displays, configure as needed the options on the tabs as needed.

   The tabs that are available depend on the voice port and call functionality policy option types that you selected.

   • Trunk Group options—For a description of these options, see the "Trunk Group Options for Voice Ports" table.

     If any trunk groups are already configured for other voice cards, they appear in the trunk groups table on this page. To edit a configured trunk group, click …, and click its pencil icon. Edit the options in the window that pops up as described in the "Trunk Group Options for Voice Ports" table, and click Save Changes. To delete a trunk group, click …, and click its trash can icon.

     After you click Save Trunk Group when saving trunk group options, configure the priority for a trunk group by double-click the Priority field for a trunk group in the Trunk Group table, entering a priority number, and pressing Enter or clicking outside of the Priority field. Valid priority numbers are integers 1 through 64. The number you enter is the priority of the POTS dial peer in the trunk group for incoming and outgoing calls.

   • Translation Profile options—For a description of these options, see the "Translation Profile Options for Calling and Called Numbers" table.

     After you click Finish when configuring translation profile options, perform these actions:

     1. Add another translation profile if needed. You can create up to two translation profiles for this endpoint.

     2. Click Save Translation Profile.

     3. For each translation profile that you create, double-click the dash (-) that displays in Direction column in the table of translation rules and select Incoming or Outgoing from the drop-down list that displays. The Incoming selection applies the corresponding translation rule to traffic that is incoming to this endpoint. The Outgoing selection applies the corresponding translation rule to traffic that is outgoing from this endpoint.

   • Station ID options—For a description of these options, see the "Station ID Options" table.

   • Line Params options—For a description of these options, see the "Line Params Options" table.

   • Tuning Params options—For a description of these options, see the "Tuning Params Options" table.

   • Supervisory Disconnect options—For a description of these options, see the "Supervisory Disconnect Options" table.

     You can configure as many supervisory disconnect events as needed.

   • DID Timers options—For a description of these options, see the "DID Timers Options" table

7. Click Next

8. In Policy Profile Name, enter a name for this child policy.

9. In Policy Profile Description, enter a description for this child policy.

10. Click Save.

When you configure POTS Dial Peers for a voice policy, you configure options that define how the system augments and manipulates calls for the POTS dial peer endpoint type.

You can configure the following options:

• Trunk Groups—The following table describes these options.

  Table 19. Trunk Group Options for POTS Dial Peers

  Option	Description	Cisco IOS CLI Equivalent
  Add New Trunk Group	Click to add a trunk group for the selected card. You can add one trunk group for a voice port.	—
  Copy from Existing	Click to copy an existing trunk group to a new trunk group. In the box that appears, change the name if desired, select a trunk group, and click Copy. A trunk group name whose name is preceded with “{Master}” is already associated with this voice policy. When you copy a this type of trunk group, the system reuses the existing trunk group without creating another instance of the trunk group definition. In this case, you cannot change the name.	—
  Name	Name of the trunk group. The name can contain up to 32 characters.	trunk group name
  Hunt-Scheme	Select the hunt scheme in the trunk group for outgoing calls: least-idle both—Searches for an idle channel with the shortest idle time least-idle even—Searches for an idle even-numbered channel with the shortest idle time least-idle odd—Searches for an idle odd-numbered channel with the shortest idle time least-used both—Searches for a trunk group member that has the highest number of available channels (applies to only PRI ISDN cards) least-used even—Searches for a trunk group member that has the highest number of available even-numbered channels (applies only to PRI ISDN cards) least-used odd—Searches for a trunk group member that has the highest number of available odd-numbered channels (applies only to PRI ISDN cards) longest-idle both—Searches for an idle odd-numbered channel with the longest idle time longest-idle even—Searches for an idle channel that has the highest number of available even-numbered channels longest-idle odd—Searches for an idle channel that has the highest number of available odd-numbered channels round-robin both—Searches trunk group members in turn for an idle channel, starting with the trunk group member that follows the last used member round-robin even—Searches trunk group member in turn for an idle even-numbered channel, starting with the trunk group member that follows the last used member round-robin odd—Searches trunk group member in turn for an idle odd-numbered channel, starting with the trunk group member that follows the last used member sequential-both—Searches for an idle channel, starting with the trunk group member with the highest preference within the trunk group sequential-even—Searches for an idle even-numbered channel, starting with the trunk group member with the highest preference within the trunk group sequential-odd—Searches for an idle odd-numbered channel, starting with the trunk group member with the highest preference within the trunk group random—Searches for a trunk group member at random and selects a channel from the member at random Default: least-used both	trunk group name hunt-scheme least-idle [both | even | odd ] hunt-scheme least-used [both | even | odd ] hunt-scheme longest-idle [both | even | odd ] hunt-scheme round-robin [both | even | odd ] hunt-scheme sequential [both | even | odd ] hunt-scheme random
  Max Calls	Enter the maximum number of calls that are allowed for the trunk group. If you do not enter a value, there is no limit on the number of calls. If the maximum number of calls is reached, the trunk group becomes unavailable for more calls. In field—Enter the maximum number of incoming calls that are allowed for this trunk group. Out field— Enter the maximum number of outgoing calls that are allowed for this trunk group. Valid range for both fields: integers 0 through 1000.	trunk group name max-calls voice number-of-calls direction [ in | out]
  Max-Retry	Select the maximum number of outgoing call attempts that the trunk group makes if an outgoing call fails. If you do not enter a value and a call fails, the system does not attempt to make the call again. Valid range: integers 1 through 5.	trunk group name max-retry attempts

• Translation Profiles—The following table describes these options.

  Table 20. Translation Profile Options for POTS Dial Peers

  Option	Description	Cisco IOS CLI Equivalent
  Add New Translation Profile	Click to add a translation profile for the selected POTS dial peer. You can create up to two translation profiles for this endpoint.	—
  Copy from Existing	Click to copy an existing translation profile to a new translation profile. In the box that appears, change the name if desired, select a called translation rule and a calling translation rule, and click Copy.	—
  Name	Name of the translation profile. The name can contain up to 32 characters.	voice translation-profile name
  Calling	Click to configure translation rules for the number that is calling in. The Translation Rules pane displays.	translate calling translation-rule-number
  Called	Click to configure translation rules for the number that is being called. The Translation Rules pane displays.	translate called translation-rule-number
  Translation Rules pane	Click Add New to create a translation rule. Alternatively, you can click Copy From Existing to copy an existing translation rule to a new translation rule. In the box that appears, change the name if desired, select a called translation rule and a calling translation rule, and click Copy. In the Translation Rule Number field, enter a unique number that designates the precedence for this rule. Valid range: integers 1 through 100. (Optional) To copy existing translation rules from a CSV file, click Import. Continue to add rules or click Finish. For detailed information about this file, see . Click Add Rule. In the Match field, enter the string that you want the translation rule to affect. Enter the string in regular expression format beginning and ending with a slash (/). For example, /^9/. To include the backslash character (\) in a match string, precede the backslash with a backslash. From the Action drop-down list, select the action that the system performs for calls that match the string in the Match field. The Reject option causes the system to reject the call. The Replace option causes the system to replace the match number with a value that you specify. If you select the Replace action, in the Replace field that displays, enter the string to which to translate the matched string. Enter the number in regular expression format beginning and ending with a slash (/). For example, //, which indicates a replacement of no string. To include the backslash character (\) in a replace string, precede the backslash with a backslash. As an example, if you specify a match string of /^9/ and a replace string of //, the system removes the leading 9 from calls with a number that begins with 9. In this case, the system translates 914085551212 to 14085551212. Click Save. Add more translation rules as needed. (Optional) Click Export to save the translation rules that you created in a CSV file. Click Finish at the bottom of the pane.	voice translation-rule number Match and Replace Rule: rule precedence /match-pattern/ / replace-pattern/ Reject Rule: rule precedence reject /match-pattern/

To configure POTS dial peers for a voice policy:

1. From the Cisco SD-WAN Manager menu, choose Configuration > Unified Communications

2. Click Add Voice Policy, and choose POTS Dial Peer in the left pane.

3. From the Add POTS Dial Peer Policy Profile drop-down list, select Create New.

   Alternatively, you can select Copy from Existing to copy an existing POTS dial peer policy to a new one. In the box that appears, select the name of the policy profile to copy, enter a new name for the profile if desired, and click Copy.

4. Select the types of POTS dial peers that you that you want to configure from the list of options that displays, and click next.

   Options are Trunk Group (beginning with Cisco IOS XE Catalyst SD-WAN Release 17.3.1a) and Translation Profile.

5. To configure trunk groups, perform the following actions.

   If any trunk groups are already configured, they appear in the trunk groups table on this page. To edit a configured trunk group, click …, and click its pencil icon. Edit the options in the window that pops up as described in the "Trunk Groups for POTS Dial Peers Options" table, and click Save Changes. To delete a trunk group, click …, and click its trash can icon.

   1. Configure trunk group options as described in the "Trunk Groups Options for POTS Dial Peers " table.

   2. Add another trunk group if needed.

      You can create up to 64 trunk groups for this endpoint.

   3. Click Save Trunk Group.

   4. Configure the priority for a trunk group by double-click the Priority field for a trunk group in the Trunk Group table, entering a priority number, and pressing Enter or clicking outside of the Priority field. Valid priority numbers are integers 1 through 64. Repeat this process for the other trunk groups in the table. The number you enter is the priority of the POTS dial peer in the trunk group for incoming and outgoing calls.

6. To configure translation profiles, perform these actions:

   1. Configure translation profile options as described in the "Translation Profile Options for POTS Dial Peers" table.

   2. Add another translation profile if needed.

      You can create up to two translation profiles for this endpoint.

   3. Click Save Translation Profile.

   4. For each translation profile that you create, double-click the dash (-) that displays in Direction column in the table of translation rules and select Incoming or Outgoing from the drop-down list that displays.

      The Incoming selection applies the corresponding translation rule to traffic that is incoming to this endpoint. The Outgoing selection applies the corresponding translation rule to traffic that is outgoing from this endpoint.

7. Click Next.

8. In Policy Profile Name, enter a name for this child policy.

9. In Policy Profile Description, enter a description for this child policy.

10. Click Save.

When you configure SIP Dial Peers for a voice policy, you configure options that define how the system augments and manipulates calls for the SIP dial peer endpoint type.

You can configure the following options, depending on the policy type for which you are configuring SIP dial peers:

• Translation Profiles—Use these options to configure translation rules for called and calling numbers on SIP dial peers. The following table describes these options.

  Table 21. Translation Profile Options for Calling Numbers on SIP Dial Peers

  Option	Description	Cisco IOS CLI Equivalent
  Add New Translation Profile	Click to add a translation profile for the selected SIP dial peer. You can create up to two translation profiles for this endpoint.	voice translation-profile name
  Copy from Existing	Click to copy an existing translation profile to a new translation profile. In the box that appears, change the name if desired, select a called translation rule and a calling translation rule, and click Copy.	—
  Calling	Click to configure translation rules for the number that is calling in. The Translation Rules pane displays.	translate calling translation-rule-number
  Called	Click to configure translation rules for the number that is being called. The Translation Rules pane displays.	translate called translation-rule-number
  Translation Rules pane	Click Add New to create a translation rule. Alternatively, you can click Copy From Existing to copy an existing translation rule to a new translation rule. In the box that appears, change the name if desired, select a called translation rule and a calling translation rule, and click Copy. In the Translation Rule Number field, enter a unique number that designates the precedence for this rule. Valid range: integers 1 through 100. (Optional) To copy existing translation rules from a CSV file, click Import. Continue to add rules or click Finish. For detailed information about this file, see . Click Add Rule. In the Match field, enter the string that you want the translation rule to affect. Enter the string in regular expression format beginning and ending with a slash (/). For example, /^9/. To include the backslash character (\) in a match string, precede the backslash with a backslash. From the Action drop-down list, select the action that the system performs for calls that match the string in the Match field. The Reject option causes the system to reject the call. The Replace option causes the system to replace the match number with a value that you specify. If you select the Replace action, in the Replace field that displays, enter the string to which to translate the matched string. Enter the number in regular expression format beginning and ending with a slash (/). For example, //, which indicates a replacement of no string. To include the backslash character (\) in a replace string, precede the backslash with a backslash. As an example, if you specify a match string of /^9/ and a replace string of //, the system removes the leading 9 from calls with a number that begins with 9. In this case, the system translates 914085551212 to 14085551212. Click Save. Add more translation rules as needed. (Optional) Click Export to save the translation rules that you created in a CSV file. Click Finish at the bottom of the pane.	voice translation-rule number Match and Replace Rule: rule precedence /match-pattern/ / replace-pattern/ Reject Rule: rule precedence reject /match-pattern/

• Media Profiles—Use these options to configure codecs to be available for the SIP trunk communication with remote dial peers and DTMF relay options to use for SIP calls. The following table describes these options.

  Table 22. Media Profile Options

  Option	Description	Cisco IOS CLI Equivalent
  Add New Media Profile	Click to add a translation profile for the dial peer.	—
  Copy from Existing	Click to copy an existing media profile to a new media profile. In the box that appears, enter a media profile number for the profile, and click Copy.	—
  Media Profile Number	Enter a number for this SIP media profile. Valid range: Integers 1 through 10000.	voice class codec tag-number
  Codec	Move from the Source list to the Target list the codecs that you want to be made available for the SIP trunk to use when communicating with the remote dial peer. Codecs in the target list are in descending order of priority, with the highest priority at the top of the list. Drag and drop items in this list to rearrange them.	voice class codec tag-number codec preference value codec-type
  DTMF	Move from the Source list to the Target list the DTMF relay options that you want the system to use for SIP calls. Items in the Target list are in descending order of priority, with the highest priority at the top of the list. Drag and drop items in this list to rearrange them. If you want to include the Inband option in the Target list, it can be the only option in that list. If you want to include other options in the Target list, move the Inband option to the Source list before saving the media profile.	dtmf-relay {[[sip-notify] [sip-kpml] [rtp-nte]]}
  Save	Click to save the configuration settings that you made.	—

• Modem Pass-through—Use these options to configure the modem pass-through feature for a SIP dial peer endpoint. The following table describes these options.

  Table 23. Modem Pass-Through Options

  Option	Description	Cisco IOS CLI Equivalent
  Add New Modem Pass-through	Click to add a modem pass-through for this SIP dial peer endpoint.	—
  Copy from Existing	Click to copy an existing modem pass-through to a new modem pass-through profile. In the box that appears, select an existing modem pass-through, enter new name if desired, and click Copy.	—
  Name	Name of the modem pass-through. This name is used when you copy an existing modem pass-through profile to a new one.	—
  Protocol	Select the protocol for the modem pass-through: None—Modem pass-through is disabled on the device NSE G.711ulaw—Uses named signaling events (NSEs) to communicate G.711ulaw codec switchover between gateways NSE G.711alaw—Uses named signaling events (NSEs) to communicate G.711alaw codec switchover between gateways	None: no modem passthrough NSE G.711 ulaw: modem passthrough nse codec g711ulaw NSE G.711 alaw: modem passthrough nse codec g711alaw
  Save Modem Pass-Through	Click to save the configuration settings that you made.	—

• Fax Protocol—Use these options to configure the fax protocol capability for a SIP dial peer endpoint. The following table describes these options.

  Table 24. Fax Protocol Options

  Option	Description	Cisco IOS CLI Equivalent
  Add New Fax Protocol	Click to add a fax protocol for the dial peer.	—
  Copy from Existing	Click to copy an existing fax protocol to a new fax protocol. In the box that appears, select an existing fax protocol, enter new name if desired, and click Copy.	—
  Name	Name of the fax protocol. This name is used when you copy an existing fax profile to a new fax profile.	—
  Primary	Select from a set of fax protocol options. Each option is a bundled set of related fax commands. For a detailed description of each bundle, see the “Primary Fax Protocol Command Bundles” table The descriptions of the bundles include the following components: nse—Uses NSEs to switch to T.38 fax relay mode force—Unconditionally uses Cisco Network Services Engines (NSE) to switch to T.38 fax relay version—Specifies a version for configuring fax speed: 0—Configures version 0, which uses T.38 version 0 (1998–G3 faxing) 3—Configures version 3, which uses T.38 version 3 (2004–V.34 or SG3 faxing) none—No fax pass-through or T.38 fax relay is attempted Pass-through—The fax stream uses one of the following high-bandwidth codecs: g711ulaw—Uses the G.711 ulaw codec g711alaw—Uses the G.711 alaw codec	fax protocol { none | pass-through {g711ulaw | g711alaw} [fallback none ]| t38 [nse [force]] [version {0 | 3}] [ls-redundancy value [hs-redundancy value]] [fallback {none | pass-through {g711ulaw | g711alaw}}]}
  Fallback	Available when the primary protocol bundle name that you selected in the Primary field begins with “T.38” or with “Fax Pass-through.” Select the fallback mode for fax transmissions. This fallback mode is used if the primary fax protocol cannot be negotiated between device endpoints. For a detailed description of each option, see the "Fallback Protocol Options” table.	fax protocol {none | pass-through {g711ulaw | g711alaw} [fallback none] | t38 [nse [force]] [version {0 | 3}] [ls-redundancy value [hs-redundancy value]] [fallback {none | pass-through {g711ulaw | g711alaw}}]}
  Low Speed	Available when the primary protocol bundle name that you selected in the Primary field begins with “T.38.” Specifies the number of redundant T.38 fax packets to be sent for the low-speed V.21-based T.30 fax machine protocol. Range: varies from 0 (no redundancy) to 5. Default: 0.	ls-redundancy value
  High Speed	Available when the primary protocol bundle name that you selected in the Primary field begins with “T.38.” Specifies the number of redundant T.38 fax packets to be sent for high-speed V.17, V.27, and V.29 T.4 or T.6 fax machine image data. Range: varies from 0 (no redundancy) to 2. Default: 0	hs-redundancy value
  Save Fax Protocol	Click to save the configuration settings that you made.	—

The following table describes the bundled sets of fax commands that are available for the Primary option when you configure the fax protocol capability for a SIP dial peer endpoint.

For low speed (ls) redundancy, the range varies from 0 (no redundancy) to 5. For high speed (HS redundancy, the range varies from 0 (no redundancy) to 2.

The following table describes the selections that are available for the Fallback option when you configure the fax protocol capability for a SIP dial peer endpoint.

To configure SIP dial peers for a voice policy:

1. From the Cisco SD-WAN Manager menu, choose Configuration > Unified Communications.

2. Click SIP Dial Peer.

3. From the Add SIP Dial Peer Policy Profile drop-down list, choose Create New.

   Alternatively, you can select Copy from Existing to copy an existing SIP dial peer policy to a new one. In the box that appears, select the name of the policy profile to copy, enter a new name for the profile if desired, and click Copy.

4. Select the policy types that you want to create and click Next:

   • Translation Profile—Lets you configure translation rules for calling and called numbers.

   • Media Profile—Lets you configure codecs to be available for the SIPtrunk communication with remote dial peers and DTMF relay options to use for SIP calls.

   • Modem Pass-through—Lets you configure the modem pass-through feature for a SIP dial peer endpoint.

   • Fax Protocol—Lets you lets you configure the fax protocol capability for a SIP dial peer endpoint. This capability is advertised and used when negotiating capabilities with the remote dial peer.

5. In the page that displays, configure options in the tabs that the following tables describe as needed.

   The tabs that are available depend on the policy types that you selected.

   • Translation Profile options—For a description of these options, see the "Translation Profile Options for Calling Numbers on SIP Dial Peers" table.

     After you click Finish when configuring a translation profile, perform these actions:

     1. Add another translation profile if needed. You can create up to two translation profiles for this endpoint.

     2. Click Save Translation Profile.

     3. For each translation profile that you create, double-click the dash (-) that displays in Direction column in the table of translation rules and select Incoming or Outgoing from the drop-down list that displays. The Incoming selection applies the corresponding translation rule to traffic that is incoming to this endpoint. The Outgoing selection applies the corresponding translation rule to traffic that is outgoing from this endpoint.

   • Media Profile options—For a description of these options, see the "Media Profile Options" table.

   • Modem Pass-through options—For a description of these options, see the "Modem Pass-Through Options" table.

   • Fax Protocol options—For a description of these options, see the "Fax Protocol Options" table.

6. Click Next.

7. In Policy Profile Name, enter a name for this child policy.

8. In Policy Profile Description, enter a description for this child policy.

9. Click Save.

When you configure SRST Phones for a voice policy, you configure options that define how the system augments and manipulates calls for the Cisco Unified SRST phone endpoint type.

The following table describes options for configuring SRST phones for a voice policy.

To configure SRST phones for a voice policy, follow these steps:

1. From the Cisco SD-WAN Manager menu, choose Configuration > Unified Communications

2. Click Add Voice Policy, and choose SRST Phone.

3. From the Add SRST Phone Policy Profile drop-down list, select Create New.

   Alternatively, you can select Copy from Existing to copy an existing policy to a new one. In the box that appears, select the name of the policy profile to copy, enter a new name for the profile if desired, and click Copy.

4. Click Media Profile, and click Next.

5. Click Add New Media Profile.

6. In the page that displays, configure options as described in the "SRST Phones Configuration Options" table.

7. Click Next.

8. In Policy Profile Name, enter a name for this child policy.

9. In Policy Profile Description, enter a description for this child policy.

10. Click Save.