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timing delay-duration through type (voice)
- timing delay-duration
- timing delay-start
- timing delay-voice tdm
- timing delay-with-integrity
- timing dialout-delay
- timing dial-pulse min-delay
- timing digit
- timing guard-out
- timing hangover
- timing hookflash-input
- timing hookflash-output
- timing ignore m-lead
- timing interdigit
- timing opx-ringwait
- timing percentbreak
- timing pulse
- timing pulse-interdigit
- timing sup-disconnect
- timing wait-wink
- timing wink-duration
- timing wink-wait
- tls
- toggle-between-two-calls
- token-root-name
- tone busytone
- tone dialtone
- tone incoming
- tone incoming system
- tone ringback alert-no-PI
- transfer
- translate
- translate (translation profiles)
- translate-outgoing
- translation-profile (dial peer)
- translation-profile (source group)
- translation-profile (trunk group)
- translation-profile (voice port)
- translation-profile (voice service POTS)
- translation-rule
- transport
- transport switch
- trunk group (global)
- trunk-group (CAS custom)
- trunkgroup (dial peer)
- trunk-group (interface)
- trunk-group (voice port)
- trunk-group-label (dial peer)
- trunk-group-label (voice source group)
- trustpoint (DSP farm profile)
- ttl
- type (settlement)
- type (voice)
timing delay-duration
To specify the delay signal duration for a specified voice port, use the timing delay-durationcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call direction for the timing delay-duration command is out. This command is supported on E&M ports only.
Examples
The following example sets the delay signal duration on a voice port to 3000 milliseconds:
voice-port 1/0/0 timing delay-duration 3000Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing delay-start
To specify the minimum delay time from outgoing seizure to out-dial address for a specified voice port, use the timing delay-startcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call direction for the timing delay-start command is out. It is supported on E&M ports only.
Examples
The following example sets the delay-start duration on a voice port to 250 milliseconds:
voice-port 1/0/0 timing delay-start 250Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing delay-voice tdm
To specify the delay after which voice packets are played out, use the timing delay-voice tdm command in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The timing delay-voice tdm command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). To avoid voice loss at the receiving end of an LMR system, use this command to configure a delay for the voice packet equal to the sum of the durations of all the injected tones and pauses configured with the inject tone command and the inject pause command.
Examples
The following example configures a timing delay of 470 milliseconds before the voice packet is played out:
voice class tone-signal mytones inject tone 1 1950 3 150 inject tone 2 2000 0 60 inject pause 3 60 inject tone 4 2175 3 150 inject tone 5 1000 0 50 voice-port 1/0/0 voice-class tone-signal mytones timing delay-voice tdm 470Note that the delay of 470 milliseconds is equal to the sum of the durations of the injected tones and pauses in the tone-signal voice class.
timing delay-with-integrity
To specify the duration of the wink pulse for the delay dial for a specified voice port, use the timing delay-with-integrity command invoice-port configuration mode. To reset to the default, use the no form of this command.
Examples
The following example sets the duration of the wink pulse for the delay dial to 10 milliseconds:
voice-port 1/0/0 timing delay-with-integrity 10Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing dialout-delay
To specify the dial-out delay for the sending digit on a specified voice port, use the timing dialout-delay command invoice-port configuration mode. To reset to the default, use the no form of this command.
Examples
The following example sets the dial-out delay to 350 milliseconds:
voice-port 1/0/0 timing dialout-delay 350Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing dial-pulse min-delay
To specify the time between wink-like pulses for a specified voice port, use the timing dial-pulse min-delaycommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Use the timing dial-pulse min-delay command with PBXs that require a wink-like pulse, even though they have been configured for delay-dial signaling. If the value for this argument is set to 0, the router does not generate this wink-like pulse. The call signal direction for this command is in.
Examples
The following example sets the time between the generation of wink-like pulses on a voice port to 350 milliseconds:
voice-port 1/0/0 timing dial-pulse min-delay 350Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing digit
To specify the dual tone multifrequency (DTMF) digit signal duration for a specified voice port, use the timing digitcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call signal direction for the timing digit command is out. This command is supported on Foreign Exchange Office (FXO), Foreign Exchange Station (FXS), and E&M ports.
Examples
The following example sets the DTMF digit signal duration on a voice port to 50 milliseconds:
voice-port 1/0/0 timing digit 50Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing guard-out
To specify the guard-out duration of a Foreign Exchange Office (FXO) voice port, use the timing guard-out command invoice-port configuration mode. To reset to the default, use the no form of this command.
timing hangover
To specify the number of milliseconds of delay before the digital signal processor (DSP) tells Cisco IOS software to turn off the E-lead after the DSP detects that the voice stream has stopped, use the timing hangover command in voice-port configuration mode. To return to the default value, use the no form of this command.
Usage Guidelines
The timing hangovercommand has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). If the voice port has been configured with the lmr e-lead voice command, use the timing hangover command to adjust the timing if the E-lead is being turned on and off too frequently.
timing hookflash-input
To specify the maximum duration of an on-hook condition that will be interpreted as a hookflash by the Cisco IOS software, use the timing hookflash-input command in voice-port configuration mode. To restore the default duration for hookflash timing, use the no form of this command.
Command Default
milliseconds : 480 milliseconds for E&M voice ports, 1000 milliseconds for FXS voice ports.
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 3600 series.
12.3(7)T
Lower limit of the range for E&M voice ports was extended to 0 milliseconds.
12.3(14)T
This command was implemented on the Cisco 2800 series and Cisco 3800 series.
12.4(2)T
This command was integrated into Cisco IOS Release 12.4(2)T.
Usage Guidelines
This command is applied to E&M or Foreign Exchange Station (FXS) interfaces.
For Land Mobile Radio E&M voice ports, the timing hookflash-input command configures the delay between when the M-lead is raised and when voice is transmitted. Setting the hookflash duration to 0 milliseconds specifies no delay in the audio input and eliminates front-end clipping.
Analog phones connected to FXS ports use hookflash to access a second dial tone to initiate some phone features, such as transfer and conference. Hookflash is an on-hook condition of short duration that is usually generated when a phone user presses the Flash button on a phone. Cisco voice gateways measure the duration of detected on-hook conditions to determine whether they should be interpreted as hookflash or not. The duration for the on-hook conditions generated by Flash buttons on phones varies for different phone types and is interpreted by Cisco IOS software as follows:
- An on-hook condition that lasts for a time period that falls inside the hookflash duration range is considered a hookflash.
- An on-hook condition that lasts for a shorter period than the lower limit of the range is ignored.
- An on-hook condition that lasts for a longer period than the higher limit of the range is considered a disconnect.
The hookflash duration range for FXS voice ports is defined as follows:
- The lower limit of the range is set in software at 150 ms, although there is also a hardware-imposed lower limit that is typically about 20 ms, depending on platform type. An on-hook condition that lasts for a shorter time than this hardware-imposed lower limit is simply not reported to the Cisco IOS software.
- The upper limit of the range is set in software at 1000 ms by default, although this value can be changed using the timing hookflash-input command in voice-port configuration mode on the voice gateway. The upper limit can be set to any value from 50 to 1550 ms. For more information, see the explanations in the "Examples" section.
This command does not affect whether hookflash relay is enabled; hookflash relay is enabled only when the dtmf-relay h245-signal command is configured on the applicable VoIP dial peers. When the dtmf-relay h245-signal command is configured, the H.323 gateway relays hookflash by using an H.245 "signal" User Input Indication method. Hookflash is sent only when an H.245 signal is available.
Examples
The following example sets an upper limit of 200 milliseconds for the hookflash duration range:
voice-port 1/0/0 timing hookflash-input 200If the timing hookflash-input command is set to X, a value greater than 150, then any on-hook duration between 150 and X is interpreted as a hookflash. For example, if X is 1550, the hookflash duration range is 150 to 1550 ms. An on-hook signal that lasts for 1250 ms is interpreted as a hookflash, but an on-hook signal of 55 ms is ignored.
voice-port 1/0/0 timing hookflash-input 1550If the timing hookflash-input command is set to X, a value less than 150, then any on-hook duration between Y, the hardware lower limit, and X is interpreted as a hookflash. For example, if X is 65, the hookflash duration range is Y to 65 ms. An on-hook signal that lasts for 1250 ms is interpreted as a disconnect, but an on-hook signal of 55 ms is interpreted as a hookflash. (This example assumes that Y for the voice gateway is lower than 55 ms.)
voice-port 1/0/0 timing hookflash-input 65timing hookflash-output
To specify the duration of hookflash indications that the gateway generates on a Foreign Exchange Office (FXO) interface, use the timing hookflash-output command in voice-port configuration mode. To restore the default duration for hookflash timing, use the no form of this command.
Command History
Release
Modification
12.1(1)T
This command was introduced on Cisco 2500, Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(4)T
Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on Cisco AS5850.
Usage Guidelines
This command does not affect whether hookflash relay is enabled; hookflash relay is enabled only when the dtmf-relay h245-signal command is configured on the applicable VoIP dial peers. Hookflash is relayed by using an H.245-signal indication and can be sent only when an H.245 signal is available.
Use the timing hookflash-outputcommand on FXO interfaces to specify the duration (in milliseconds) of a hookflash indication. To set hookflash timing parameters for analog voice interfaces, use the timing command.
timing ignore m-lead
To ignore M-lead or voice activity detection (VAD) changes for a specified amount of time after sending the E-lead off signal, use the timing ignore m-lead command in voice-port configuration mode. To return to the default value, use the no form of this command.
Usage Guidelines
- The timing ignore m-lead command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Use this command to reduce echo feedback on an LMR voice port. This command has an effect only if the voice port is configured for half duplex mode.
timing interdigit
To specify the dual-tone multifrequency (DTMF) interdigit duration for a specified voice port, use the timing interdigitcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call signal direction for the timing interdigit command is out. This command is supported on Foreign Exchange Office (FXO), Foreign Exchange Station (FXS), and E&M ports.
Examples
The following example sets the DTMF interdigit duration on a voice port to 150 milliseconds:
voice-port 1/0/0 timing interdigit 150Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing opx-ringwait
To set the maximum wait time for detecting the next ring on FXO ports, use the timing opx-ringwaitcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
This command prevents the voice gateway from prematurely disconnecting private line automatic ring-down (PLAR) off-premises extension (OPX) calls when the duration between ring tones from the switch is more than 6 sec. The absence of a ring tone from the switch indicates that the originating party has disconnected the call. Because some analog switches take longer than 6 sec to generate the ring tone, the voice gateway could clear the call leg while it is still ringing for a PLAR OPX call, unless the 6-sec default is changed with this command.
timing percentbreak
To specify the percentage of the break period for dialing pulses for a voice port, use the timing percentbreak command invoice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The timing percentbreakcommand is supported on Foreign Exchange Office (FXO) and E&M voice ports only.
timing pulse
To specify the pulse dialing rate for a specified voice port, use the timing pulsecommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call signal direction for the timing pulse command is out. This command is supported on Foreign Exchange Office (FXO) and E&M ports only.
Examples
The following example sets the pulse dialing rate on a voice port to 15 pulses per second:
voice-port 1/0/0 timing pulse 15Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing pulse-interdigit
To specify the pulse interdigit timing for a specified voice port, use the timing pulse-interdigitcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call signal direction for the timing pulse-interdigit command is out. This command is supported on Foreign Exchange Office (FXO) and E&M ports only.
Examples
The following example sets the pulse-dialing interdigit timing on a voice port to 300 milliseconds:
voice-port 1/0/0 timing pulse-interdigit 300Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing sup-disconnect
To define the minimum time 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), use the timing sup-disconnect command in voice-port configuration mode. To restore the default value, use the no form of this command.
Command Default
The default minimum time is 350 milliseconds before a supervisory disconnect occurs.
Usage Guidelines
Prior to the implementation of the timing sup-disconnect command, analog Foreign Exchange Office (FXO) ports could not detect short disconnect signals lasting fewer than 350 ms in duration. Using this command, you can specify a wait period from 50 to 1500 ms to ensure that when an on-hook indication persists for a time that is longer than the configured value, the on-hook condition is considered intentional and a hang-up is signaled on the POTS call leg.
This timer affects only analog loop-start FXO voice ports.
Even though the timing sup-disconnect command can be entered under the voice port in FXO ground-start signaling, the changes in the timer setting take effect only in FXO loop-start signaling.
timing wait-wink
To set the maximum time to wait for wink signal after an outgoing seizure is sent, use the timing wait-winkcommand in voice port configuration mode. To restore the default value, use the no form of this command.
Examples
The following example configures the maximum time to wait for wink signaling after an outgoing seizure is sent on a voice port for 300 milliseconds:
voice-port 1/0/0 timing wait-wink 300Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dial-out delay for the sending digit on a specified voice port.
timing delay-with-integrity
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing wink-duration
To specify the timing for transmit and receive wink-signal duration for a voice port, use the timing wink-durationcommand in voice-port configuration mode. To reset to the default values, use the no form of this command.
Syntax Description
time
Maximum transmit duration, in milliseconds (ms), for a wink-start signal. The range is from 50to 3000. The default is 200.
receive
Indicates that a range is to be specified for a received wink-start signal.
minimum
Received minimum wink length, in milliseconds. The range is from 40 to 2950. The default is 140.
maximum
Received maximum wink length, in milliseconds. The range is from 150 to 3150. The default is 290.
Command Default
Transmit wink-duration timing is set to 200 ms. The received wink-duration timing minimum is set to 140 ms and the maximum is set to 290 ms.
Command History
Release
Modification
11.3(1)T
This command was introduced on Cisco 3600 series.
11.3(1)MA
This command was integrated into Cisco IOS Release 11.3(1)MA and support was added for the Cisco MC3810.
12.4(13)
This command was integrated into Cisco IOS Release 12.4(13) and the receive keyword and minimum and maximum arguments were added.
Usage Guidelines
The call signal direction for the timing wink-duration command is out. This command is supported on ear and mouth (E&M) ports only.
When wink-start signaling is used, the originating side seizes the line by going off-hook and then waits for an acknowledgment from the other end before initiating a call. The acknowledgment is a reversal of polarity (off-hook) for a timing period referred to as a wink. A wink should occur no earlier than 100 ms after the receipt of the incoming seizure signal. In addition to the signaling function, the wink start serves as an integrity check that identifies a malfunctioning trunk and allows the network to send a reorder tone to the calling party.
When you set the receive range, the minimum and maximum values of acceptable wink must provide an acceptable range of at least 50 ms. For example, entering the command timing wink-duration receive 160 200 results in an error message.
Examples
The following example shows how to set the transmit wink-signal duration on voice port 1/0/0 to 300 ms:
voice-port 1/0/0 timing wink-duration 300The following example shows how to set the range for the receive wink-signal duration on voice port 1/0/0 to 160 to 210 ms:
voice-port 1/0/0 timing wink-duration receive 160 210Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing delay-with-integrity
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-wait
Specifies the maximum wink-wait duration for a specified voice port.
timing wink-wait
To specify the maximum wink-wait duration for a specified voice port, use the timing wink-waitcommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
The call signal direction for the timing wink-wait command is out. This command is supported on ear and mouth (E&M) ports only.
Examples
The following example sets the wink-wait duration on a voice port to 300 milliseconds:
voice-port 1/0/0 timing wink-wait 300Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
timeouts wait-release
Configures the timeout value for releasing voice ports.
timing clear-wait
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
timing delay-duration
Specifies the delay signal duration for a specified voice port.
timing delay-start
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
timing delay-with-integrity
Specifies the duration of the wink pulse for the delay dial for a specified voice port.
timing dialout-delay
Specifies the dialout delay for the sending digit on a specified voice port.
timing dial-pulse min-delay
Specifies the time between wink-like pulses for a specified voice port.
timing digit
Specifies the DTMF digit signal duration for a specified voice port.
timing interdigit
Specifies the DTMF interdigit duration for a specified voice port.
timing percentbreak
Specifies the percentage of a break period for a dialing pulse for a specified voice port.
timing pulse
Specifies the pulse dialing rate for a specified voice port.
timing pulse-interdigit
Specifies the pulse interdigit timing for a specified voice port.
timing wink-duration
Specifies the maximum wink signal duration for a specified voice port.
tls
To enable Transport Layer Security (TLS) for the Skinny Client Control Protocol (SCCP) connection between the SCCP server and the SCCP client, use the tlscommand in DSP farm profile configuration mode. To disable secure SCCP signaling, use the no form of this command.
Usage Guidelines
Use the tlscommand to enable secure SCCP signaling exchange. The configuration can be modified only when the dspfarm profile is shut down. To shut down the dsp farm profile, configure the no shutdown command.
toggle-between-two-calls
To define a Feature Access Code (FAC) to access the Toggle Between Two Calls feature in feature mode on analog phones connected to FXS ports, use the toggle-between-two-calls command in STC application feature-mode call-control configuration mode. To return the code to its default, use the no form of this command.
Usage Guidelines
This command changes the value of the FAC for Toggle Between Two Calls from the default (#5) to the specified value.
If you attempt to configure this command with a value that is already configured for another FAC in feature mode, you receive a message. This message will not prevent you from configuring the feature code. If you configure a duplicate FAC, the system implements the first feature it matches in the order of precedence as determined by the value for each FAC (#1 to #5).
If you attempt to configure this command with a value that precludes or is precluded by another FAC in feature mode, you receive a message. If you configure a FAC to a value that precludes or is precluded by another FAC in feature mode, the system always executes the call feature with the shortest code and ignores the longer code. For example, 1 will always preclude 12 and 123. These messages will not prevent you from configuring the feature code. You must configure a new value for the precluded code in order to enable phone user access to that feature.
Examples
The following example shows how to change the value of the feature code for the Toggle Between Two Calls feature from the default (#5). With this configuration, a phone user in basic call mode presses hook flash to get the first dial tone, then dials an extension number to connect to a second call. During the second call, the user presses a hook flash to get a feature tone and then dials 55 to toggle back to the previous call party.
Router(config)# stcapp call-control mode feature Router(config-stcapp-fmcode)# toggle-between-two-calls 55 Router(config-stcapp-fmcode)# exitRelated Commands
Command
Description
conference
Defines FAC in Feature Mode to initiate a three-party conference.
drop-last-conferee
Defines FAC in feature mode to use to drop last active call during a three-party conference.
hangup-last-active-call
Defines FAC in feature mode to drop last active call during a three-party conferencee.
transfer
Defines FAC in feature mode to connect a call to a third party that the phone user dials.
token-root-name
To specify which root or Certificate Authority (CA) certificate the router uses to validate the settlement token in the incoming setup message, use the token-root-namecommand in settlement configurationmode. To reset to the default, use the no form of this command.
Examples
The following example defines the token-root-name as "sample":
token-root-name sampleThe following example shows new output for the show settlement command to display the value of the token-root-name command:
Settlement Provider 0 Operation Status = UP Type = osp Address url = https://1.14.115.100:8444/ Encryption = all (default) Token Root Name = sample Max Concurrent Connections = 20 (default) Connection Timeout = 3600 (s) (default) Response Timeout = 1 (s) (default) Retry Delay = 2 (s) (default) Retry Limit = 1 (default) Session Timeout = 86400 (s) (default) Customer Id = 1000 Device Id = 2000 Roaming = Disabled (default) Signed Token = On Number of Connections = 1 Number of Transactions = 0tone busytone
To enable automatic busytone generation in a basic call scenario, use the tone busytonecommand in dial peer voice configuration mode. To disable automatic busytone generation, use the no form of this command.
Usage Guidelines
The automatic busytone generation after remote disconnect in basic call mode feature is enabled and disabled per dial peer with the tone busytone remote-onhookcommand. The tone busytonecommand is available to all dial peer services. Each service determines whether to utilize or enable it. For STCAPP, only the Foreign eXchange Subscriber (FXS) loop-start port will enable this service.
NoteThe tone busytone command cannot coexist with thedialtone generation after remote-onhook feature. Because the tone dialtoneis a default configuration, you must disable the feature using the no tone dialtonecommand before connfiguring the tone busytonecommand.
Use the show dial-peer voice command or the show stcapp device voice command to verify the feature is enabled.
tone dialtone
To enable automatic dial-tone generation in basic call mode, use the tone dialtonecommand in dial peer configuration mode. To disable automatic dial-tone generation, use the no form of this command.
Usage Guidelines
Use this command to generate immediate dial tone once a remote party disconnects, similar to what the user experiences in a PBX environment. If you disable this feature using the no form of this command, the user is required to go on hook or perform a hookflash to generate dial tone after the remote party disconnects in a basic two-part call scenario. This feature is supported on Skinny Client Control Protocol (SCCP) gateway controlled loop-start FXS ports only.
Examples
The following examples show that the automatic Dial Tone Generation After Remote Onhook feature is enabled. Because the dial tone generation after remote onhook feature is enabled by default, it does not display in the show running-config output.
Router# show running-config service stcapp dial-peer voice 3001 pots port 1/1/1 Router# show dial-peer voice 3001 VoiceEncapPeer3001 peer type = voice, system default peer = FALSE, information type = voice, ! ! ! in bound application associated: 'stcapp' dial tone generation after remote-onhook = enabled Router# show stcapp device voice-port 1/1/1 Port Identifier: 1/1/1 ! Dialtone after remote-onhook feature: activatedThe following examples show the dial tone generation after remote onhook feature disabled.
Router# show running-config no tone dialtone remote-onhook dial-peer voice 3002 pots service stcapp port 1/1/0tone incoming
To activate 2100-Hz answer (ANS) tone detection on either the IP or the PSTN side of the network and to disable the echo suppressor, use the tone incomingcommand in voice-service VoIP configuration mode or VoIP dial-peer configuration mode. To deactivate tone detection and disable the echo suppressor, use the no form of this command.
tone incoming [ ip | pstn ] { ans-all auto-control | ans disable echo suppressor | anspr disable echo suppressor }
no tone incoming
Syntax Description
ip
(Optional) Specifies tone detection on the IP side of the network.
pstn
(Optional) Specifies tone detection on the PSTN side of the network.
ans auto-control
Detects ANS tone and enables standard actions for modem tones.
ans-all disable echo suppressor
Detects modem answer tones and disables echo suppressor.
anspr disable echo suppressor
Detects /ANS tone and disables echo suppressor.
Usage Guidelines
Use this command in voice-service VoIP or VoIP dial-peer configuration mode to activate detection of all ANS, ANSam, and ANSpr tones and enable or disable echo canceller control. When this command is issued in voice-service VoIP configuration mode, all dial peers are globally configured unless a specific dial peer is configured for no tone incoming.
To deactivate all 2100-Hz ANS, ANSam, and ANSpr tone detection on either the IP or the PSTN side of the network, and enable the echo canceller, use the no tone incoming command in voice-service VoIP configuration or VoIP dial-peer configuration mode.
If neither IP nor PSTN is specified, all ANS, ANSam, and ANSpr tones are detected on both sides of the network, and the echo suppressor is disabled in all cases.
The tone incoming ip ans-all auto-control command is equivalent to these two commands together:
The tone incoming pstn ans-all auto-control command is equivalent to these two commands together:
The tone incoming ans-all auto-control command is equivalent to these four commands together:
- tone incoming ip ans disable echo suppressor
- tone incoming ip anspr disable echo suppressor
- tone incoming pstn ans disable echo suppressor
- tone incoming pstn anspr disable echo suppressor
When modem tones from either the IP or PSTN direction are received, the echo canceller can be dynamically disabled to allow modem calls to pass through.
The IP tone detector feature applies only on the following NextPort platforms: Cisco AS5350, Cisco AS5400, and Cisco AS5850--and only with SIP and H.323 voice signaling. It does not apply to MGCP in VoIP dial-peer configuration mode.
The gateway must be configured for G.711 codecs for the IP tone detector feature to work (see the "Examples" section).
To display the status of the echo canceller, use the show port operational status command.
Examples
The following example configures tone detection of ANS tones in voice-service VoIP configuration mode:
Router(conf-voi-serv)# tone incoming ip ans disable echo supressorThe following example configures tone detection of all incoming ANS, ANSam, and ANSpr tones on a dial peer:
Router(config-dial-peer)# tone incoming ip ans-all auto-controltone incoming system
To set a dial peer for tone incoming or no tone incoming, use the tone incoming system command in VoIP dial-peer configuration mode. To block the voice service VoIP settings for a dial peer, use the no form of this command.
Usage Guidelines
Use this command in VoIP dial-peer configuration mode to activate or deactivate tone detection and to enable echo canceller control. When modem tones from either the IP or PSTN directions are received. The echo canceller can be dynamically disabled to allow modem calls through. This command is used primarily to allow or to block global voice service VoIP configuration settings.
To block the voice service VoIP settings for a dial peer, use the no tone incoming system command.
Examples
The following example shows activating tone detection for a dial peer.
Router(config-dial-peer)# tone incoming systemThe following example shows deactivating tone detection for a dial peer.
Router(config-dial-peer)# no tone incoming systemRelated Commands
Command
Description
tone incoming ans disable echo suppressor
Activates ANS tone detection.
tone incoming anspr disable echo canceller
Activates ANSpr tone detection.
tone incoming ans-all auto-control
Activates ANS, ANSam, and ANSpr tone detection.
show port operational status
Displays the status of the echo canceller.
tone ringback alert-no-PI
To generate automatic ringback for the caller when no Progress Indicator (PI) alert has been received over the H.323 network, use the tone ringback alert-no-PIcommand in dial-peer configurationmode. To disable automatic ringback, use the no form of this command.
Usage Guidelines
Use this command to generate ringback in an H.323 network when the attached device (for example, an ISDN device) cannot.
transfer
To define a Feature Access Code (FAC) to access the Call Transfer feature in feature mode on analog phones connected to FXS ports, use the transfer command in STC application feature-mode call-control configuration mode. To return the code to its default, use the no form of this command.
Usage Guidelines
This command changes the value of the FAC for Call Transfer from the default (#2) to the specified value.
If you attempt to configure this command with a value that is already configured for another FAC in feature mode, you receive a message. This message will not prevent you from configuring the feature code. If you configure a duplicate FAC, the system implements the first feature it matches in the order of precedence as determined by the value for each FAC (#1 to #5).
If you attempt to configure this command with a value that precludes or is precluded by another FAC in feature mode, you receive a message. If you configure a FAC to a value that precludes or is precluded by another FAC in feature mode, the system always executes the call feature with the shortest code and ignores the longer code. For example, 1 will always preclude 12 and 123. These messages will not prevent you from configuring the feature code. You must configure a new value for the precluded code in order to enable phone user access to that feature.
Examples
The following example shows how to change the value of the feature code for the Call Transfer feature from the default (#2). With this configuration, a phone user presses hook flash to get the first dial tone, then dials an extension number to connect to a second call. When the second call is established, the user presses hook flash to get a feature tone and then dials 22 to transfer the call; the user hears silence after the call is transferred.
Router(config)# stcapp call-control mode feature Router(config-stcapp-fmcode)# transfer 22 Router(config-stcapp-fmcode)# exitRelated Commands
Command
Description
conference
Defines FAC in Feature Mode to initiate a three-party conference.
drop-last-conferee
Defines FAC in feature mode to use to drop last active call during a three-party conference.
hangup-last-active-call
Defines FAC in feature mode to drop last active call during a three-party conferencee.
toggle-between-two-calls
Defines FAC in feature mode to toggle between two active calls.
translate
To apply a translation rule to manipulate dialed digits on an inbound POTS call leg, use the translate command in voice-port configuration mode. To remove the translation rule, use the no form of this command.
translate { calling-number | called-number } name-tag
no translate { calling-number | called-number } name-tag
Syntax Description
calling -number
Translation rule applies to the inbound calling party number.
called -number
Translation rule applies to the inbound called party number.
name -tag
Tag number by which the rule set is referenced. This is an arbitrarily chosen number. Range is from 1 to 2147483647. There is no default value.
Command History
Release
Modification
12.0(7)XR1
This command was introduced for VoIP on Cisco AS5300.
12.0(7)XK
This command was implemented for VoIP on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T and implemented for VoIP Cisco AS5300, Cisco 7200, and Cisco 7500.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
A translation rule is a general-purpose digit-manipulation mechanism that performs operations such as automatically adding telephone area and prefix codes to dialed numbers.
Examples
The following example applies translation rule 21 to the POTS inbound calling-party number:
translation-rule 21 rule 1 555.% 1408555 subscriber international rule 2 7.% 1408555 abbreviated international voice-port 0:1 translate calling-number 21The following example applies translation rule 20 to the POTS inbound called-party number:
translation-rule 20 rule 1 .%555.% 7 any abbreviated voice-port 0:1 translate called-number 20Related Commands
Command
Description
numbering-type
Specifies number type for the VoIP or POTS dial peer.
rule
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
show translation-rule
Displays the contents of all the rules that have been configured for a specific translation name.
translate-outgoing
Applies a translation rule to a calling party number or a called party number for outgoing calls.
translation-rule
Creates a translation name and enters translation-rule configuration mode.
voip-incoming translation-rule
Captures calls that originate from H.323-compatible clients.
translate (translation profiles)
To associate a translation rule with a voice translation profile, use the translate command in voice translation-profile configuration mode. To delete the translation rule from the profile, use the no form of this command.
translate { called | calling | redirect-called | redirect-target } translation-rule-number
no translate { called | calling | redirect-called | redirect-target } translation-rule-number
Syntax Description
called
Associates the translation rule with called numbers.
calling
Associates the translation rule with calling numbers.
redirect -called
Associates the translation rule with redirected called numbers.
redirect-target
Associates the translation rule with transfer-to numbers and call-forwarding final destination numbers.
translation -rule-number
Number of the translation rule to use for the call translation. Valid range is from 1 to 2147483647. There is no default value.
Command History
Release
Modification
12.0(7)XR1
This command was introduced on the Cisco AS5300.
12.0(7)XK
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T and implemented on the following platforms: Cisco 1750, Cisco AS5300, Cisco 7200 series, and Cisco 7500 series.
12.1(2)T
This command was implemented on the Cisco MC3810.
12.2(11)T
This command was reconfigured for voice translation-profile configuration mode. The redirect-calledkeyword and translation-rule-numberargument were added.
12.4(11)XJ
The redirect-target keyword was added.
12.4(15)T
The redirect-target keyword was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
Use this command as part of a voice translation-profile definition. Enter this command for each translation rule that is part of the profile definition.
Examples
The following example defines voice translation profile "sjmorning" with two translation rules: translation rule 15 for called numbers and translation rule 36 for calling numbers.
Router(config)# voice translation-profile sjmorning Router(cfg-translation-profile)# translate called 15 Router(cfg-translation-profile)# translate calling 36Related Commands
Command
Description
rule (voice translation-rule)
Sets the criteria for the translation-rule.
show voice translation-profile
Displays the configuration of the translation-profile.
translation-profile (dial-peer)
Assigns a translation profile to a dial peer.
translation-profile (source group)
Assigns a translation profile to a source IP group.
translation-profile (trunk group)
Assigns a translation profile to a trunk group.
translation-profile (voice port)
Assigns a translation profile to a voice port.
translation-profile (voice service POTS)
Assigns a translation profile to an NFAS interface.
voice translation-profile
Initiates the translation-profile definition.
voice translation-rule
Initiates the translation-rule definition.
translate-outgoing
To apply a translation rule to manipulate dialed digits on an outbound POTS or VoIP call leg, use the translate-outgoing command indial-peer configuration mode. To disable the translation rule, use the no form of this command.
translate-outgoing { calling-number | called-number } name-tag
no translate-outgoing { calling-number | called-number } name-tag
Command History
Release
Modification
12.0(7)XR1
This command was introduced for VoIP on Cisco AS5300.
12.0(7)XK
This command was implemented for VoIP on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(1)T
This command was integrated into Cisco IOS Release 12.2(1)T and implemented for VoIP on the Cisco 1750, Cisco AS5300, Cisco 7200, and Cisco 7500. support for the Cisco MC3810 is not included in this release.
12.1(2)T
This command is supported on the Cisco MC3810 in this release.
Examples
The following example applies translation rule 21 to the VoIP outbound calling number:
translation-rule 21 rule 1 555.% 1408555 subscriber international rule 2 7.% 1408555 abbreviated international dial-peer voice 100 voip translate-outgoing calling-number 21The following example applies translation rule 20 to the VoIP called number:
translation-rule 20 rule 1 .%555.% 7 any abbreviated dial-peer voice 100 voip translate-outgoing called-number 20Related Commands
Command
Description
numbering-type
Specifies number type for the VoIP or POTS dial peer.
rule
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
show translation-rule
Displays the contents of all the rules that have been configured for a specific translation name.
translate
Applies a translation rule to a calling party number or a called party number for incoming calls.
translation-rule
Creates a translation name and enters translation-rule configuration mode.
voip-incoming translation-rule
Captures calls that originate from H.323-compatible clients.
translation-profile (dial peer)
To assign a translation profile to a dial peer, use the translation-profile command in dial peer configuration mode. To delete the translation profile from the dial peer, use the no form of this command.
Usage Guidelines
Use the translation-profile command to assign a predefined translation profile to a dial peer.
Examples
The following example assigns the translation profile named "profile1" to handle translation of outgoing calls for a dial peer:
Router(config)# dial-peer voice 111 pots Router(config-dial-peer)# translation-profile outgoing profile1Related Commands
Command
Description
rule (voice translation-rule)
Sets the criteria for the translation rule.
show voice translation-profile
Displays the configuration of a translation profile.
translate (translation profiles)
Assigns a translation rule to a translation profile.
voice translation-profile
Initiates the translation-profile definition.
voice translation-rule
Initiates the translation-rule definition.
translation-profile (source group)
To assign a translation profile to a source IP group, use the translation-profile command in source group configuration mode. To delete the translation profile from the source IP group, use the no form of this command.
Usage Guidelines
Use the translation-profile command to assign a predefined translation profile to a source IP group.
Examples
The following example assigns the translation profile named "chicago" to handle translation of incoming calls for a voice source group:
Router(config)# voice source-group alpha Router(cfg-source-grp)# translation-profile incoming chicagoRelated Commands
Command
Description
rule (voice translation-rule)
Sets the criteria for the translation rule.
show voice translation-profile
Displays the configuration of a translation profile.
translate (translation profiles)
Assigns a translation rule to a translation profile.
voice translation-profile
Initiates the translation-profile definition.
voice translation-rule
Initiates the translation-rule definition.
translation-profile (trunk group)
To assign a translation profile to a trunk group, use the translation-profile command in trunk group configuration mode. To delete the translation profile from the trunk group, use the no form of this command.
Usage Guidelines
Use the translation-profile command to assign a predefined translation profile to a trunk group.
Examples
The following example assigns the translation profile named "newyork" to handle translation of incoming calls for a trunk group:
Router(config)# trunk group 10 Router(config-trunk-group)# translation-profile incoming newyorkRelated Commands
Command
Description
rule (voice translation-rule)
Sets the criteria for the translation rule.
show voice translation-profile
Displays the configuration of a translation profile.
translate (translation profiles)
Assigns a translation rule to a translation profile.
voice translation-profile
Initiates the translation-profile definition.
voice translation-rule
Initiates the translation-rule definition.
translation-profile (voice port)
To assign a translation profile to a voice port, use the translation-profile command in voice port configuration mode. To delete the translation profile from the voice port, use the no form of this command.
Usage Guidelines
Use the translation-profile command to assign a predefined translation profile to a voice port.
Examples
The following example assigns the translation profile named "chicago" to handle translation of incoming calls and a translation profile named "sanjose" to handle outgoing calls for a voice port:
Router(config)# voice-port 1/0/0 Router(config-voiceport)# translation-profile incoming chicago Router(config-voiceport)# translation-profile outgoing sanjoseRelated Commands
Command
Description
rule (voice translation-rule)
Sets the criteria for the translation rule.
show voice translation-profile
Displays the configuration of a translation profile.
translate (translation profiles)
Assigns a translation rule to a translation profile.
voice translation-profile
Initiates the translation-profile definition.
voice translation-rule
Initiates the translation-rule definition.
translation-profile (voice service POTS)
To assign a translation profile to a non-facility associated signaling (NFAS) interface, use the translation-profile command in voice service POTS configuration mode. To delete the translation profile from the interface, use the no form of this command.
translation-profile [ incoming | outgoing ] controller [ T1 | E1 ] unit-number name
no translation-profile [ incoming | outgoing ] controller [ T1 | E1 ] unit-number name
Usage Guidelines
Use the translation-profile command to assign a predefined translation profile to an NFAS interface.
Examples
The following example assigns to an NFAS interface the translation profile named "delta1" to outgoing T1 calls on controller slot 3 and translation profile "alpha" to incoming T1 calls on controller slot 2:
Router(config)# voice service pots Router(conf-voi-serv)# translation-profile outgoing controller T1 3 delta1 Router(conf-voi-serv)# translation-profile incoming controller T1 2 alphaRelated Commands
Command
Description
rule (voice translation-rule)
Sets the criteria for the translation rule.
show voice translation-profile
Displays the configuration of a translation profile.
translate (translation profiles)
Assigns a translation rule to a translation profile.
voice translation-profile
Initiates the translation-profile definition.
voice translation-rule
Initiates the translation-rule definition.
translation-rule
To create a translation name and enter translation-rule configuration mode to apply rules to the translation name, use the translation-rule command in global configuration mode. To disable the translation rule, use the no form of this command.
Command History
Examples
The following example creates translation rule 21 and applies a rule to it:
translation-rule 21 rule 1 555.% 1408555 subscriber internationalRelated Commands
Command
Description
numbering-type
Specifies number type for the VoIP or POTS dial peer.
rule
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
test translation-rule
Tests the execution of the translation rules on a specific name tag.
translate
Applies a translation rule to a calling party number or a called party number for incoming calls.
translate-outgoing
Applies a translation rule to a calling party number or a called party number for outgoing calls.
voip-incoming translation-rule
Captures calls that originate from H.323-compatible clients.
transport
To configure the Session Inititation Protocol (SIP) user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP, Transport Layer Security (TLS) over TCP, or User Datagram Protocol (UDP) socket, use the transportcommand in SIP user agent configuration mode. To block reception of SIP signaling messages on a particular socket, use the no form of this command.
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300 platforms.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
12.2(2)XA
This command was implemented on Cisco AS5400 and Cisco AS5350 platforms.
12.2(2)XB1
This command was implemented on Cisco AS5850 platforms.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms were not included in this release.
12.2(11)T
Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.
12.4(6)T
The optional tls keyword was added to the command.
Usage Guidelines
This command controls whether messages reach the SIP service provider interface (SPI). Setting tcp, tls over tcp, or udpas the protocol causes this to be the protocol for which SIP user agents listen on port 5060.
To block reception of SIP signaling messages on a specific socket, use the no form of this command.
To reset this command to the default value, use the default command.
Examples
The following example sets the SIP user agent to allow the reception of SIP signaling messages on the UDP socket:
sip-ua transport udpThe following example sets the SIP user agent to allow the reception of SIP signaling messages on the TCP socket:
sip-ua transport tcpThe following example sets the SIP user agent to allow the reception of SIP signaling messages on the TLS over TCP socket:
sip-ua transport tcp tlstransport switch
To enable switching between UDP and TCP transport mechanisms globally for large Session Initiation Protocol (SIP) messages, use the transport switchcommand in SIP configuration mode. To disable switching between UDP and TCP transport mechanisms globally for large SIP messages, use the no form of this command.
Usage Guidelines
Switching between transports is provided globally on the router and also on an individual VoIP dial peer.
- Dial-peer mode. You can configure transport for a specific dial peer by using the voice-class sip transport switchcommand. The voice-class sip transport switch command in dial-peer configuration mode takes precedence over the transport switchcommand in global configuration mode.
- SIP mode. You can configure transport globally by using the transport switchcommand. The transport switch command is considered only when there is no matching VoIP dial peer.
In a call forking scenario, if this command is configured, the configuration applies to all forks.
Examples
The following example enables switching of the transport from UDP to TCP:
Router(config)# voice service voip Router(config-voi-srv)# sip Router(conf-serv-sip)# transport switch udp tcpRelated Commands
Command
Description
debug ccsip transport
Enables tracing of the SIP transport handler and the TCP or UDP process.
sip
Enters SIP configuration mode from voice-service VoIP configuration mode.
voice -class sip transport switch
Enables switching between transport mechanisms if the SIP message is larger than 1300 bytes for a specific dial peer.
trunk group (global)
To define or modify the definition of a trunk group and to enter trunk group configuration mode, use the trunk groupcommand in global configuration mode. To delete the trunk group, use the no form of this command.
Usage Guidelines
Use the trunk group command to assign a number or a name to a set of trunk characteristics. The set of characteristics, or profile , is assigned to specific trunks as part of the usual trunk configuration steps.
The trunk group command initiates the profile definition and switches from global configuration to trunk group configuration mode. Additional commands are available to construct the characteristics of the profile.
Up to 1000 trunk groups can be configured on the gateway provided that the gateway has sufficient memory to store the profiles. If you see the message "Trunk group name could not be added as the threshold has been reached", enter the debug tgrmcommand and check the number of trunk groups or check for insufficient memory.
To associate a trunk group with an interface, use the trunk-group (interface) command. A trunk group that was created using the trunk group (global) command can be associated with an interface. However, a trunk group need not be defined globally before being associated with an interface. If a trunk group has not been defined globally, it will be created by issuing the trunk-group (interface) command.
Examples
The following example creates trunk group 5 and configures the trunk group profile:
Router(config)# trunk group 5 Router(config-trunk-group)# carrier-id allcalls Router(config-trunk-group)# max-calls voice 500 in Router(config-trunk-group)# hunt-scheme round-robin even up Router(config-trunk-group)# translation-profile incoming 3 Router(config-trunk-group)# translation-profile outgoing 2 Router(config-trunk-group)# exitThe following example creates a trunk group named "mytrunk" and configures the trunk group profile:
Router(config)# trunk group mytrunk Router(config-trunk-group)# carrier-id local Router(config-trunk-group)# max-calls voice 500 Router(config-trunk-group)# hunt-scheme least-idle Router(config-trunk-group)# translation-profile incoming 1 Router(config-trunk-group)# translation-profile outgoing 12 Router(config-trunk-group)# exitRelated Commands
Command
Description
carrier-id (trunk group)
Identifies the carrier that owns the trunk group.
description (trunk group)
Permits a description to be associated with a trunk group.
hunt-scheme least-idle
Specifies the least-idle channel search method for incoming and outgoing calls.
hunt-scheme least-used
Specifies the least-used channel search method for incoming and outgoing calls.
hunt-scheme longest-idle
Specifies the longest-idle channel search method for incoming and outgoing calls.
hunt-scheme random
Specifies the random channel search method for incoming and outgoing calls.
hunt-scheme round-robin
Specifies the round-robin channel search method for incoming and outgoing calls.
hunt-scheme sequential
Specifies the sequential channel search method for incoming and outgoing calls.
max-calls
Specifies the number of incoming and outgoing voice and data calls that a trunk group can handle.
show trunk group
Displays the configuration of trunk groups.
translation-profile (trunk group)
Defines call number translation profiles for incoming and outgoing calls.
trunk-group (interface)
Assigns an ISDN PRI or NFAS interface to a trunk group.
trunk-group (CAS custom)
To assign a channel-associated signaling (CAS) trunk to a trunk group, use the trunk-groupcommand in CAS custom configuration mode. To delete the CAS trunk from the trunk group, use the no form of this command.
Usage Guidelines
Use the trunk-group command to assign a CAS trunk as a member of a trunk group. This assignment provides the CAS trunk with carrier information, a hunt scheme for finding an available channel for the outgoing call, and translation profiles for number translation.
If more than one CAS trunk is assigned to the same trunk group, the preference-num value determines the order in which the trunk group uses the interfaces. A preference-num value of 1 is the highest preference so that the trunk is used first; a value of 64 is the lowest preference so that the trunk is used last. If no value is entered for preference-num, the software assigns the trunk a preference of 65, which causes that trunk to be used after all other trunks are used.
If two CAS trunks have the same preference-num, the trunk that was configured first is used before the other trunk.
A CAS trunk can belong to only one trunk group.
If an interface is removed from the CAS trunk, the interface is removed automatically from the trunk group. A new nonprimary CAS interface is automatically a member of the same trunk group as its primary CAS interface.
Examples
The following example assigns two CAS interfaces to trunk group "westcoast". The preference value for DS0 group 2 is lower than for DS0 group 1; hence DS0 group 2 has a higher priority. Trunk group "westcoast" uses DS0 group 2 first.
Router(config)# controller T1 1/0 Router(config-controller)# ds0-group 1 timeslots 1-10 type e&m-fgd Router(config-controller)# cas-custom 1 Router(config-controller)# trunk-group westcoast 5 Router(config-controller)# exit Router(config)# controller T1 1/0 Router(config-controller)# ds0-group 2 timeslots 15-20 type e&m-fgd Router(config-controller)# cas-custom 2 Router(config-controller)# trunk-group westcoast 3 Router(config-controller)# exittrunkgroup (dial peer)
To assign a dial peer to a trunk group for trunk group label routing, use the trunkgroupcommand in dial-peer configuration mode. To delete the dial peer from the trunk group, use the no form of this command.
Usage Guidelines
Use the trunkgroup command to assign an outgoing dial peer as a member of one or more trunk groups. This assignment provides the dial peer with carrier information, a hunt scheme for finding an available channel for the outgoing call, and translation profiles for number translation.
If the dial peer is a member of more than one trunk group, use the preference-num value to set the order in which the trunk groups will be used for the dial peer. A preference-num value of 1 is the highest preference so that the trunk group is used first; a value of 64 is the lowest preference so that the trunk group is used last. If no value is entered for preference-num, the software assigns the trunk group a preference of 65, which causes that trunk group to be selected after all other trunks are used.
If two trunk groups have the same preference-num, the trunk group that was configured first is used before the other trunk group.
Examples
In the following example, dial peer 112 should use the trunk group "east17" and trunk group "north5" for outbound dial peer matching. When selecting a trunk group, "north5" is used first because it has a higher preference than "east17":
Router(config)# dial-peer voice 112 pots Router(config-dial-peer)# trunkgroup east17 3 Router(config-dial-peer)# trunkgroup north5 1trunk-group (interface)
To assign an ISDN PRI or Non-Facility Associated Signaling (NFAS) interface to a trunk group, use the trunk-groupcommand in interface configuration mode. To delete the interface from the trunk group, use the no form of this command.
Usage Guidelines
Use the trunk-group command to configure an ISDN PRI or Non-Facility Associated Signaling (NFAS) interface as a member of a trunk group. This assignment provides the interface with carrier information, a hunt scheme for finding an available channel for the outgoing call, and translation profiles for number translation.
If more than one interface is assigned to the same trunk group, the preference_num value determines the order in which the trunk group uses the interfaces. A preference-num value of 1 is the highest preference so that the interface is used first; a value of 64 is the lowest preference so that the interface is used last. If no value is entered for preference-num, the software assigns the interface a preference of 65, which causes that interface to be selected after all other interfaces are used.
If two interfaces have the same preference-num, the interface that was configured first is used before the other interface.
An interface can belong to only one trunk group. Multiple interfaces can belong to the same trunk group.
If an NFAS interface group is assigned as a member of a trunk group, all the subinterfaces belong to that trunk group.
If a subinterface is removed from the NFAS group, the subinterface is removed automatically from the trunk group.
If a new nonprimary NFAS interface is added to the NFAS group, that interface automatically becomes a member of the same trunk group as its primary NFAS interface.
Examples
The following example assigns an ISDN interface to trunk group "eastern" with a preference of 3.
Router(config)# interface Serial2:23 Router(config-if)# no ip address Router(config-if)# isdn switch-type primary-ni Router(config-if)# isdn T306 30000 Router(config-if)# isdn T310 10000 Router(config-if)# no cdp enable Router(config-if)# trunk-group eastern 3 Router(config-if)# exitIf another interface were assigned to trunk group "eastern" with preference of 1 or 2, the trunk group would use that interface before the one shown above.
trunk-group (voice port)
To assign an analog voice port to a trunk group, use the trunk-groupcommand in voice port configuration mode. To delete the trunk group, use the no form of this command.
Usage Guidelines
Use the trunk-group command to configure an analog voice port as a member of a trunk group. This assignment provides the voice port with carrier information, a hunt scheme for finding an available channel for the outgoing call, and translation profiles for number translation.
If more than one voice port is assigned to the same trunk group, the preference-num value determines the order by which the trunk group uses the voice ports. A preference-num value of 1 is the highest preference so that the voice port is used first; a value of 64 is the lowest preference so that the voice port is used last. If no value is entered for preference-num, the software assigns the voice port a preference of 65, which causes that voice port to be selected after all other voice ports are used.
If two voice ports have the same preference-num, the voice port that was configured first is used before the other voice port.
A voice port can belong to only one trunk group. Multiple voice ports can belong to the same trunk group.
Examples
The following example assigns voice port 1/0/0 and voice port 1/0/1 to trunk group "north5". Trunk group "north5" uses voice port 1/0/1 before using voice port 1/0/0 because voice port 1/0/1 has preference 1, which is a higher priority than voice port 1/0/0, with preference 2.
Router(config)# voice port 1/0/0 Router(config-voiceport)# translation-profile incoming 7 Router(config-voiceport)# translation-profile outgoing 4 Router(config-voiceport)# trunk-group north5 2 Router(config-voiceport)# exit Router(config)# voice port 1/0/1 Router(config-voiceport)# translation-profile incoming 3 Router(config-voiceport)# translation-profile outgoing 8 Router(config-voiceport)# trunk-group north5 1 Router(config-voiceport)# exittrunk-group-label (dial peer)
To specify a trunk group as the source or target of a call, use the trunk-group-labelcommand in dial peer configuration mode. To delete the trunk group label, use the no form of this command.
Usage Guidelines
An originating gateway uses the source trunk group label as a matching key to route the call over an inbound dial peer. The terminating gateway uses the target trunk group label to select a dial peer for routing the outbound call over a POTS line.
If a dial peer has a source (or target) carrier ID already defined, then assigning a source (or target) trunk group label to that same dial peer overrides the source (or target) carrier ID. The same is true for the reverse: if a dial peer has a source (or target) trunk group label defined, then assigning a source (or target) carrier ID for that same dial peer overrides the source (or target) trunk group label.
The name of a trunk group label and carrier ID cannot be the same in dial peers.
Examples
The following example shows that dial peer 112 should use trunk group label "north3" for inbound dial peer matching and trunk group label "east17" for outbound dial peer matching:
Router(config)# dial-peer voice 112 pots Router(config-dial-peer)# trunk-group-label source north3 Router(config-dial-peer)# trunk-group-label target east17trunk-group-label (voice source group)
To define a trunk group label in a source IP group, use the trunk-group-labelcommand in voice source group configuration mode. To delete the trunk group label, use the no form of this command.
Usage Guidelines
A terminating gateway uses the source trunk group label as a search key to find a source IP group for the incoming VoIP call. The gateway uses the target trunk group label to select an outbound dial peer to route the call over a POTS line.
If a source IP group has a source (or target) carrier ID already defined, then assigning a source (or target) trunk group label to that same source IP group overrides the source (or target) carrier ID. The same is true for the reverse: if a source IP group has a source (or target) trunk group label defined, then assigning a source (or target) carrier ID for that same source IP group overrides the source (or target) trunk group label.
The name of a trunk group label and carrier ID of the same type (source or target) cannot be the same in the source IP group.
Examples
The following example shows that source IP group "alpha" uses trunk group "north3" to search for a source IP group for incoming VoIP calls and trunk group "east17" for outbound dial peer matching:
Router(config)# voice source-group alpha Router(cfg-source-grp)# trunk-group-label source north3 Router(cfg-source-grp)# trunk-group-label target east17trustpoint (DSP farm profile)
To associate a trustpoint with a DSP farm profile, use the trustpoint command in DSP farm profile configuration configuration mode. To remove the association, use the no form of this command.
Usage Guidelines
Use this command to associate trustpoints with secure DSP farm profiles only. Use the security keyword of the dspfarm profile command to configure a secure DSP farm profile. If the trustpoint is not already configured, you are prompted to configure the trustpoint.
ttl
To set the expiration timer for advertisements, enter the ttlcommand in Annex G configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Usage Guidelines
The address templates or routes that are static to this Annex G border element (BE) can be advertised to its neighbors. A time-to-live (TTL) value is associated with each of the advertised routes. The TTL value indicates how long the neighbor should consider the routes valid. On expiration of the ttl, the neighbor must query the addressing information again.
type (settlement)
To point to the provider type and the specific settlement server, use the type command in settlement configuration mode. To disable this command, use the no form of this command.
Usage Guidelines
This command defines the settlement server that is doing the accounting and enables the server to do the accounting.
Examples
The following example enables authentication of VoIP calls to the PSTN using a single settlement server:
settlement 0 type uni-ospRelated Commands
Command
Description
connection -timeout
Sets the connection timeout.
customer -id
Sets the customer identification.
device -id
Sets the device identification.
encryption
Specifies the encryption method.
max -connection
Sets the maximum simultaneous connections.
response -timeout
Sets the response timeout.
retry -delay
Sets the retry delay.
retry -limit
Sets the connection retry limit.
session -timeout
Sets the session timeout.
settlement
Enters settlement configuration mode.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no shutdown
Brings up the settlement provider and then shuts it down.
url
Specifies the Internet service provider (ISP) address.
type (voice)
To specify the E&M interface type, use the type command in voice-port configuration mode. To reset to the default, use the no form of this command.
Syntax Description
Usage Guidelines
Use the type command to specify the E&M interface for a particular voice port. With 1, the tie-line equipment generates the E-signal to the PBX type grounding the E-lead. The tie-line equipment detects the M-signal by detecting current flow to ground. If you select 1, a common ground must exist between the line equipment and the PBX.
With 2, the interface requires no common ground between the equipment, thereby avoiding ground loop noise problems. The E-signal is generated toward the PBX by connecting it to SG. The M-signal is indicated by the PBX connecting it to SB. While Type 2 interfaces do not require a common ground, they do have the tendency to inject noise into the audio paths because they are asymmetrical with respect to the current flow between devices.
NoteE&M Type 4 is not a supported option. However, Type 4 operates similarly to Type 2 except for the M-lead operation. On Type 4, the M-lead states are open/ground, compared to Type 2, which is open/battery. Type 4 can interface with Type 2. To use Type 4 you can set the E&M voice port to Type 2 and perform the necessary M-lead rewiring.
With 3, the interface operates the same as Type 1 interfaces with respect to the E-signal. The M-signal, however, is indicated by the PBX connecting it to SB on assertion and alternately connecting it to SG during inactivity. If you select 3, a common ground must be shared between equipment.
With 5, the Type 5 line equipment indicates E-signal to the PBX by grounding the E-lead. The PBX indicates M-signal by grounding the M-lead. A Type 5 interface is quasi-symmetrical in that while the line is up, current flow is more or less equal between the PBX and the line equipment, but noise injection is a problem.
