- A
- B
- cac master through call application stats
- call application voice through call denial
- call fallback through called-number (dial peer)
- caller-id (dial peer) through ccm-manager switchover-to-backup
- ccs connect (controller) through clear vsp statistics
- clid through credentials (sip-ua)
- default (auto-config application) through direct-inward-dial
- disable-early-media through dualtone
- E
- F
- G
- H
- icpif through irq global-request
- isdn bind-l3 through ixi transport http
- K
- L
- map q850-cause through mgcp package-capability
- mgcp persistent through mmoip aaa send-id secondary
- mode (ATM/T1/E1 controller) through mwi-server
- N
- O
- package through pattern
- periodic-report interval through proxy h323
- Q
- R
- sccp through service-type call-check
- session through sgcp tse payload
- show aal2 profile through show call filter match-list
- show call history fax through show debug condition
- show dial-peer through show gatekeeper zone prefix
- show gateway through show modem relay statistics
- show mrcp client session active through show sip dhcp
- show sip service through show trunk hdlc
- show vdev through show voice statistics memory-usage
- show voice trace through shutdown (voice-port)
- signal through srv version
- ss7 mtp2-variant through switchover method
- target carrier-id through timeout tsmax
- timeouts call-disconnect through timing clear-wait
- timing delay-duration through type (voice)
- U
- vad (dial peer) through voice-class sip encap clear-channel
- voice-class sip error-code-override through vxml version 2.0
- W
- Z
- 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
- 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-duration command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing delay-duration time
no timing delay-duration time
Syntax Description
time |
Delay signal duration for delay dial signaling, in milliseconds. Range is from 100 to 5000. The default is 2000. |
Command Default
2000 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series. |
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 3000
Related Commands
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-start command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing delay-start time
no timing delay-start
Syntax Description
time |
Minimum delay time, in milliseconds, from outgoing seizure to outdial address. Range is from 20 to 2000. The default on the Cisco 3600 series is 300. |
Command Default
Cisco 3600 series: 300 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series routers. |
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 250
Related Commands
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.
timing delay-voice tdm milliseconds
no timing delay-voice tdm milliseconds
Syntax Description
milliseconds |
Duration, in milliseconds, of the timing delay. Range is integers from 1 to 1500. Default is 0. |
Command Default
milliseconds: 0 milliseconds
Command Modes
Voice-port configuration
Command History
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 470
Note 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.
Related Commands
|
|
|
|---|---|
inject pause |
Specifies a pause between injected tones. |
inject tone |
Specifies a wakeup or frequency selection tone to be played out before the voice packet. |
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 in voice-port configuration mode. To reset to the default, use the no form of this command.
timing delay-with-integrity time
no timing delay-with-integrity
Syntax Description
time |
Duration of the wink pulse for the delay dial, in milliseconds. Range is from 0 to 5000. The default is 0. |
Command Default
0 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)MA |
This command was introduced on the Cisco MC3810. |
Usage Guidelines
This command is supported on E&M ports only.
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 10
Related Commands
timing dialout-delay
To specify the dial-out delay for the sending digit on a specified voice port, use the timing dialout-delay command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing dialout-delay time
no timing dialout-delay time
Syntax Description
time |
Dial-out delay, in milliseconds, for the sending digit or cut-through on a Foreign Exchange Office (FXO) trunk or an E&M immediate trunk. Range is from 100 to 5000. The default is 300. |
Command Default
300 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)MA |
This command was introduced on Cisco MC3810. |
Examples
The following example sets the dial-out delay to 350 milliseconds:
voice-port 1/0/0
timing dialout-delay 350
Related Commands
timing dial-pulse min-delay
To specify the time between wink-like pulses for a specified voice port, use the timing dial-pulse min-delay command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing dial-pulse min-delay time
no timing dial-pulse min-delay
Syntax Description
time |
Time between wink-like pulses, in milliseconds. Range is from 0 to 5000. The default is 300. |
Command Default
300 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series. |
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 350
Related Commands
timing digit
To specify the dual tone multifrequency (DTMF) digit signal duration for a specified voice port, use the timing digit command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing digit time
no timing digit
Syntax Description
time |
The DTMF digit signal duration, in milliseconds. Range is 5 from 0 to 100. The default is 100. |
Command Default
100 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series. |
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 50
Related Commands
timing guard-out
To specify the guard-out duration of a Foreign Exchange Office (FXO) voice port, use the timing guard-out command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing guard-out time
no timing guard-out
Syntax Description
time |
Duration of the guard-out period, in milliseconds. The range is from 300 to 3000. The default is 2000. |
Defaults
The default is 2000 milliseconds
Command Modes
Voice-port configuration
Command History
Usage Guidelines
This command is supported on FXO voice ports only.
For Caller ID to work for FXO ports registered to a Cisco Unified CM, the range in milliseconds must be between 1000 to 2000.
Examples
The following example sets the timing guard-out duration on a voice port to 1000 milliseconds:
voice-port 1/0/0
timing guard-out 1000
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.
timing hangover milliseconds
no timing hangover milliseconds
Syntax Description
milliseconds |
The number of milliseconds for which the E-lead stays active after VAD determines that the voice stream has stopped. Valid values are 0 to 10000. The default is 250 milliseconds. |
Command Default
milliseconds: 250 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
12.3(4)XD |
This command was introduced. |
12.3(7)T |
This command was integrated into Cisco IOS Release 12.3(7)T. |
Usage Guidelines
The timing hangover 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). 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.
Examples
The following example configures E-lead on voice port 1/0/1 on a Cisco 3745 to stay active for 300 milliseconds after VAD determines that the voice stream has stopped:
voice-port 1/0/1
timing hangover 300
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.
timing hookflash-input milliseconds
no timing hookflash-input
Syntax Description
Command Default
milliseconds: 480 milliseconds for E&M voice ports, 1000 milliseconds for FXS voice ports.
Command Modes
Voice-port configuration
Command History
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 200
If 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 1550
If 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 65
Related Commands
|
|
|
dtmf-relay (Voice over IP) |
Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network. |
timing 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.
timing hookflash-output time
no timing hookflash-output
Syntax Description
time |
Duration of the hookflash, in milliseconds. Range is from 50 to 1550. The default is 400 milliseconds. |
Command Default
400 milliseconds
Command Modes
Voice-port configuration
Command History
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-output command 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.
Examples
The following example implements timing for the hookflash with a duration of 200 milliseconds.
voice-port 1/0/0
timing hookflash-output 200
Related Commands
|
|
|
dtmf-relay (Voice over IP) |
Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network. |
voice-port |
Enters voice-port configuration mode. |
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.
timing ignore m-lead milliseconds
no timing ignore m-lead milliseconds
Syntax Description
milliseconds |
The number of milliseconds following the sending of the E-lead off signal for which the M-lead and VAD changes are ignored. Valid values are 0 to 10000. The default is 0 milliseconds. |
Command Default
milliseconds: 0 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
12.3(4)XD |
This command was introduced. |
12.3(7)T |
This command was integrated into Cisco IOS Release 12.3(7)T. |
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.
Examples
The following example configures voice port 1/0/1 on a Cisco 3745 to ignore M-lead or VAD changes for 500 milliseconds after sending the E-lead off signal:
voice-port 1/0/1
timing ignore m-lead 500
timing interdigit
To specify the dual-tone multifrequency (DTMF) interdigit duration for a specified voice port, use the timing interdigit command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing interdigit time
no timing interdigit time
Syntax Description
time |
DTMF interdigit duration, in milliseconds. Range is from 50 to 500. The default is 100. |
Command Default
100 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series. |
11.3(1)MA |
This command was supported on Cisco MC3810. |
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 150
Related Commands
timing opx-ringwait
To set the maximum wait time for detecting the next ring on FXO ports, use the timing opx-ringwait command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing opx-ringwait msecs
no timing opx-ringwait
Syntax Description
msecs |
Maximum duration, in milliseconds, to wait for the next ring. Range is 2000 to 10000. Default is 6000. |
Command Default
Timeout for detecting ring tones is 6000 ms (6 sec).
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
12.4(4)T |
This command was introduced. |
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.
Examples
The following example sets the timeout for the next ring to 8 sec:
voice-port 2/0/10
timing opx-ringwait 8000
Related Commands
|
|
|
|---|---|
voice-port |
Enters voice-port configuration mode. |
show voice port |
Displays configuration information about a specific voice port. |
timing percentbreak
To specify the percentage of the break period for dialing pulses for a voice port, use the timing percentbreak command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing percentbreak percent
no timing percentbreak
Syntax Description
percent |
Percentage of the break period for dialing pulses. Range is from 20 to 80. The default is 50. |
Command Default
50 percent
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The timing percentbreak command is supported on Foreign Exchange Office (FXO) and E&M voice ports only.
Examples
The following example sets the break period percentage on a voice port to 30 percent:
voice-port 0/0/1
timing percentbreak 30
Related Commands
|
|
|
timing pulse |
Configures the pulse dialing rate for a voice port. |
timing pulse-interdigit |
Configures the pulse interdigit timing for a voice port. |
timing pulse
To specify the pulse dialing rate for a specified voice port, use the timing pulse command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing pulse pulses-per-second
no timing pulse pulses-per-second
Syntax Description
pulses-per-second |
Pulse dialing rate, in pulses per second. Range is from 10 to 20. The default is 20. |
Command Default
20 pulses per seconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on the Cisco 3600 series. |
11.3(1)MA |
This command was supported on the Cisco MC3810. |
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 15
Related Commands
timing pulse-interdigit
To specify the pulse interdigit timing for a specified voice port, use the timing pulse-interdigit command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing pulse-interdigit time
no timing pulse-interdigit time
Syntax Description
time |
Pulse dialing interdigit timing, in milliseconds. Range is from 100 to 1000. The default is 500. |
Command Default
500 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series. |
11.3(1)MA |
This command was supported on Cisco MC3810. |
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 300
Related Commands
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.
timing sup-disconnect milliseconds
no timing sup-disconnect milliseconds
Syntax Description
Command Default
The default minimum time is 350 milliseconds before a supervisory disconnect occurs.
Command Modes
Voice-port configuration
Command History
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.
Examples
The following example sets the timer to wait 500 ms after detecting an on-hook signal before a supervisory disconnect occurs on the POTS call leg:
voice-port 1/0/0
timing sup-disconnect 500
Related Commands
|
|
|
|---|---|
show voice port |
Displays configuration information about a specific voice port. |
voice-port |
Enters voice-port configuration mode. |
timing wait-wink
To set the maximum time to wait for wink signal after an outgoing seizure is sent, use the timing wait-wink command in voice port configuration mode. To restore the default value, use the no form of this command.
timing wait-wink milliseconds
no timing wait-wink milliseconds
Syntax Description
milliseconds |
Maximum time to wait for wink signal after an outgoing seizure is sent. Valid entries are from 100 to 6500 milliseconds (ms). Supported on ear and mouth (E&M) ports only. |
Defaults
milliseconds: 550 milliseconds
Command Modes
Voice port configuration
Command History
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 300
Related Commands
timing wink-duration
To specify the timing for transmit and receive wink-signal duration for a voice port, use the timing wink-duration command in voice-port configuration mode. To reset to the default values, use the no form of this command.
timing wink-duration {time | receive minimum maximum}
no timing wink-duration
Syntax Description
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 Modes
Voice-port configuration
Command History
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 300
The 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 160 210
Related Commands
timing wink-wait
To specify the maximum wink-wait duration for a specified voice port, use the timing wink-wait command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing wink-wait time
no timing wink-wait
Syntax Description
time |
Maximum wink-wait duration, in milliseconds, for a wink start signal. Range is from 100 to 6500. The default is 200. |
Defaults
200 milliseconds
Command Modes
Voice-port configuration
Command History
|
|
|
|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series. |
11.3(1)MA |
This command was supported on Cisco MC3810. |
12.4(12) |
The millisecond range was extended from 5000 to 6500. |
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 300
Related Commands
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 tls command in DSP farm profile configuration mode. To disable secure SCCP signaling, use the no form of this command.
tls
no tls
Syntax Description
This command has no arguments or keywords.
Command Default
Secure SCCP signaling exchange is enabled by default.
Command Modes
DSP farm profile configuration (config-dspfarm-profile #)
Command History
|
|
|
|---|---|
12.4(22)YB |
This command was introduced. |
Usage Guidelines
Use the tls command 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.
Examples
The following example shows how to configure the tls command to enable TLS support for digital signal processor (DSP) farm services profile 1:
Router(config)# dspfarm profile 1 transcode security
Router(config-dspfarm-profile)# tls
Related Commands
|
|
|
|---|---|
dspfarm profile |
Enters DSP farm profile configuration mode and defines a profile for DSP farm services. |
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.
toggle-between-two-calls keypad-character
no toggle-between-two-calls
Syntax Description
keypad-character |
Character string of one to four characters that can be dialed on a telephone keypad (0—9, *, #). Default is #5. |
Command Default
The default value is #5.
Command Modes
STC application feature-mode call-control configuration (config-stcapp-fmcode)
Command History
|
|
|
|---|---|
15.0(1)M |
This command was introduced. |
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)# exit
Related Commands
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-name command in settlement configuration mode. To reset to the default, use the no form of this command.
token-root-name name
no token-root-name
Syntax Description
name |
Certificate identification name as configured with the crypto ca identity name command or the crypto ca trusted-root name command. |
Command Default
The terminating gateway uses the CA certificate to validate the settlement token.
Command Modes
Settlement configuration
Command History
|
|
|
|---|---|
12.1(1)T |
This command was introduced on Cisco 2600 series, Cisco 3600 series, Cisco AS5300, and Cisco AS5800. |
Examples
The following example defines the token-root-name as "sample":
token-root-name sample
The 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 = 0
Related Commands
tone busytone
To enable automatic busytone generation in a basic call scenario, use the tone busytone command in dial peer voice configuration mode. To disable automatic busytone generation, use the no form of this command.
tone busytone remote-onhook
no tone busytone remote-onhook
Syntax Description
remote-onhook |
Generates busy tone after remote onhook in basic call mode. |
Command Default
Automatic busytone generation after remote disconnect is disabled.
Command Modes
Dial peer voice configuration (config-dial-peer)
Command History
|
|
|
|---|---|
12.4(20)T |
This command was introduced. |
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-onhook command. The tone busytone command 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.
Note The tone busytone command cannot coexist with the dialtone generation after remote-onhook feature. Because the tone dialtone is a default configuration, you must disable the feature using the no tone dialtone command before connfiguring the tone busytone command.
Use the show dial-peer voice command or the show stcapp device voice command to verify the feature is enabled.
Examples
The following example shows busytone generation after remote disconnect being configured:
Router(config-dial-peer)# tone busytone remote-onhook
Related Commands
tone dialtone
To enable automatic dial-tone generation in basic call mode, use the tone dialtone command in dial peer configuration mode. To disable automatic dial-tone generation, use the no form of this command.
tone dialtone remote-onhook
no tone dialtone remote-onhook
Syntax Description
remote-onhook |
Generates dial tone after remote onhook in basic call mode. |
Command Default
Automatic dial-tone generation after remote disconnect is enabled.
Command Modes
Dial peer configuration (config-dial-peer)
Command History
|
|
|
|---|---|
12.4(6)XE |
This command was introduced. |
12.4(11)T |
This command was integrated into Cisco IOS Release 12.4(11)T. |
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: activated
The 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 1/1/0
Related Commands
tone 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 incoming command in voice-service VoIP configuration mode or 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
Command Default
Tone incoming detection is not enabled.
Command Modes
Voice-service VoIP configuration
Dial peer configuration
Command History
|
|
|
|---|---|
12.3(14)T |
This command was introduced. |
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 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:
•tone incoming ip ans disable echo suppressor
•tone incoming ip anspr disable echo suppressor
The tone incoming pstn ans-all auto-control command is equivalent to these two commands together:
•tone incoming pstn ans disable echo suppressor
•tone incoming pstn anspr disable echo suppressor
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 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 supressor
The 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-control
Related Commands
|
|
|
|---|---|
tone incoming system |
Sets a dial peer for tone incoming or no tone incoming detection. |
show port operational status |
Displays the status of the echo canceller. |
tone 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.
tone incoming system
no tone incoming system
Syntax Description
This command has no arguments or keywords.
Command Default
The dial peer is set for tone incoming.
Command Modes
dial peer configuration
Command History
|
|
|
|---|---|
12.3(14)T |
This command was introduced. |
Usage Guidelines
Use this command in 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 system
The following example shows deactivating tone detection for a dial peer.
Router(config-dial-peer)# no tone incoming system
Related Commands
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-PI command in dial peer configuration mode. To disable automatic ringback, use the no form of this command.
tone ringback alert-no-PI
no tone ringback alert-no-PI
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
Usage Guidelines
The tone ringback alert-no-PI command is used to generate ringback in an H.323 network when the attached device (for example, an ISDN device) cannot.
Examples
The following example activates ringback for a VoIP dial peer numbered 322:
router(config)# dial-peer voice 322 voip
router(config-dial-peer)# tone ringback alert-no-PI
Related Commands
|
|
|
|---|---|
progress_ind |
Sets a specific PI in call Setup, Progress, or Connect messages from an H.323 VoIP gateway. |
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.
transfer keypad-character
no transfer
Syntax Description
keypad-character |
Character string of one to four characters that can be dialed on a telephone keypad (0—9, *, #). Default is #2. |
Command Default
The default value is #2.
Command Modes
STC application feature-mode call-control configuration (config-stcapp-fmcode)
Command History
|
|
|
|---|---|
15.0(1)M |
This command was introduced. |
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)# exit
Related Commands
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
Command Default
No default behavior or values
Command Modes
Voice-port configuration
Command History
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 21
The 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 20
Related Commands
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
Command Default
No translation rule is associated with the translation profile.
Command Modes
Voice translation-profile configuration (cfg-translation-profile)
Command History
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 36
Related Commands
translate-outgoing
To apply a translation rule to manipulate dialed digits on an outbound POTS or VoIP call leg, use the translate-outgoing command in dial 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
Syntax Description
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
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 21
The 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 20
Related Commands
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.
translation-profile {incoming | outgoing} name
no translation-profile {incoming | outgoing} name
Syntax Description
incoming |
Specifies that this translation profile handles incoming calls. |
outgoing |
Specifies that this translation profile handles outgoing calls. |
name |
Name of the translation profile. |
Defaults
No default behavior or values
Command Modes
Dial peer configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
12.4(22)T |
Support for IPv6 was added. |
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 profile1
Related Commands
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.
translation-profile incoming name
no translation-profile incoming name
Syntax Description
incoming |
Specifies that this translation profile handles incoming calls. |
name |
Name of the translation profile. |
Command Default
No default behavior or values
Command Modes
Source group configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
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 chicago
Related Commands
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.
translation-profile {incoming | outgoing} name
no translation-profile {incoming | outgoing} name
Syntax Description
incoming |
Specifies that this translation profile handles incoming calls. |
outgoing |
Specifies that this translation profile handles outgoing calls. |
name |
Name of the translation profile. |
Command Default
No default behavior or values
Command Modes
Trunk group configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
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 newyork
Related Commands
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.
translation-profile {incoming | outgoing} name
no translation-profile {incoming | outgoing} name
Syntax Description
incoming |
Specifies that this translation profile handles incoming calls. |
outgoing |
Specifies that this translation profile handles outgoing calls. |
name |
Name of the translation profile. |
Command Default
No default behavior or values
Command Modes
Voice port configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
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 sanjose
Related Commands
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
Syntax Description
Command Default
No default behavior or values
Command Modes
Voice service POTS configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
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 alpha
Related Commands
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.
translation-rule name-tag
no translation-rule name-tag
Syntax Description
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 Default
No default behavior or values
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to all translation rules.
Examples
The following example creates translation rule 21 and applies a rule to it:
translation-rule 21
rule 1 555.% 1408555 subscriber international
Related Commands
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 transport command in SIP user agent configuration mode. To block reception of SIP signaling messages on a particular socket, use the no form of this command.
transport {tcp tls | udp}
no transport {tcp tls | udp}
Syntax Description
tcp tls |
SIP user agent receives SIP messages on TLS over TCP port 5060. |
udp |
SIP user agent receives SIP messages on UDP port 5060. |
Command Default
TCP, TLS over TCP, and UDP transport protocols are enabled.
Command Modes
SIP user-agent configuration
Command History
Usage Guidelines
This command controls whether messages reach the SIP service provider interface (SPI). Setting tls over tcp or udp as 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 no form of this 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 udp
The 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 tls
Related Commands
|
|
|
|---|---|
sip-ua |
Enables the SIP user agent configuration commands. |
transport switch
To enable switching between UDP and TCP transport mechanisms globally for large Session Initiation Protocol (SIP) messages, use the transport switch command 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.
transport switch udp tcp
no transport switch udp tcp
Syntax Description
udp |
Enables switching the transport mechanism from UDP on the basis of the size of the SIP request being greater than the MTU size. |
tcp |
Enables switching transport to TCP. |
Command Default
Disabled.
Command Modes
SIP configuration
Command History
|
|
|
|---|---|
12.3(8)T |
This command was introduced. |
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 switch command. The voice-class sip transport switch command in dial peer configuration mode takes precedence over the transport switch command in global configuration mode.
•SIP mode. You can configure transport globally by using the transport switch command. 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 tcp
Related Commands
trunk group
To define or modify the definition of a trunk group and to enter trunk group configuration mode, use the trunk group command in global configuration mode. To delete the trunk group, use the no form of this command.
trunk group name
no trunk group name
Syntax Description
name |
Name of the trunk group. Valid names contain a maximum of 63 alphanumeric characters. |
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
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 tgrm command and check the number of trunk groups or check for insufficient memory.
Examples
The following example assigns the number 5 to a trunk group profile:
Router(config)# trunk group 5
Router(config-trunk-group)# carrier-id allcalls
Router(config-trunk-group)# maxcalls 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)# exit
The following example assigns the name "newyork" to a trunk group profile:
Router(config)# trunk group newyork
Router(config-trunk-group)# carrier-id local
Router(config-trunk-group)# maxcalls 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)# exit
Related Commands
trunk-group (CAS custom)
To assign a channel-associated signaling (CAS) trunk to a trunk group, use the trunk-group command in CAS custom configuration mode. To delete the CAS trunk from the trunk group, use the no form of this command.
trunk-group name [preference-num]
no trunk-group name [preference-num]
Syntax Description
Command Default
Preference-num is set lower than 64 (internally set to 65)
Command Modes
CAS custom configuration
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
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)# exit
Related Commands
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show trunk group |
Displays the configuration of a trunk group. |
trunkgroup (dial peer)
To assign a dial peer to a trunk group for trunk group label routing, use the trunkgroup command in dial peer configuration mode. To delete the dial peer from the trunk group, use the no form of this command.
trunkgroup name preference-num
no trunkgroup name
Syntax Description
Command Default
Preference-num is set lower than 64 (internally set to 65)
Command Modes
Dial peer configuration
Command History
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|---|---|
12.1(3)T |
This command was introduced. |
12.2 |
This command was integrated into the Cisco IOS Release 12.2. |
12.2(11)T |
The preference-num argument was added. |
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 1
Related Commands
trunk-group (interface)
To assign an ISDN PRI or Non-Facility Associated Signaling (NFAS) interface to a trunk group, use the trunk-group command in interface configuration mode. To delete the interface from the trunk group, use the no form of this command.
trunk-group name [preference-num]
no trunk-group name [preference-num]
Syntax Description
Command Default
Preference-num is set lower than 64 (internally set to 65)
Command Modes
Interface configuration
Command History
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)# exit
If 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.
Related Commands
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show trunk group |
Displays the configuration of the trunk group. |
trunk-group (voice port)
To assign an analog voice port to a trunk group, use the trunk-group command in voice port configuration mode. To delete the trunk group, use the no form of this command.
trunk-group name [preference-num]
no trunk-group name [preference-num]
Syntax Description
Command Default
Preference-num is set lower than 64 (internally set to 65)
Command Modes
Voice port configuration
Command History
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|---|---|
12.2(11)T |
This command was introduced. |
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)# exit
Related Commands
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show trunk group |
Displays the configuration of a trunk group. |
trunk-group-label (dial peer)
To specify a trunk group as the source or target of a call, use the trunk-group-label command in dial peer configuration mode. To delete the trunk group label, use the no form of this command.
trunk-group-label {source | target} name
no trunk-group-label {source | target} name
Syntax Description
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
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|---|---|
12.2(11)T |
This command was introduced. |
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 east17
Related Commands
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carrier-id (dial peer) |
Specifies the carrier associated with a VoIP call. |
show dial-peer voice |
Displays configuration information for dial peers. |
trunk-group-label (voice source group)
To define a trunk group label in a source IP group, use the trunk-group-label command in voice source group configuration mode. To delete the trunk group label, use the no form of this command.
trunk-group-label {source | target} name
no trunk-group-label {source | target} name
Syntax Description
Command Default
No default behavior or values
Command Modes
Voice source group configuration
Command History
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|---|---|
12.2(11)T |
This command was introduced. |
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 east17
Related Commands
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carrier-id (dial peer) |
Specifies the carrier associated with a VoIP call. |
show voice source-group |
Displays the configuration for voice source IP groups. |
trustpoint (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.
trustpoint trustpoint-label
no trustpoint trustpoint-label
Syntax Description
trustpoint-label |
Label of the trustpoint to be associated with the digital signal processor (DSP) farm profile. |
Command Default
No trustpoints are associated with the DSP farm profile
Command Modes
DSP farm profile configuration (config-dspfarm-profile)
Command History
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|---|---|
12.4(11)XW1 |
This command was introduced. |
12.4(20)T |
This command was integrated into Cisco IOS Release 12.4(20)T. |
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.
Examples
The following example associates the trustpoint dspfarm with the DSP farm profile 101:
Router(config)# dspfarm profile 101 conference security
Router(config-dspfarm-profile)# trustpoint dspfarm
Related Commands
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dspfarm profile |
Enters DSP farm profile configuration mode and defines a profile for digital signal processor (DSP) farm services. |
ttl
To set the expiration timer for advertisements, enter the ttl command in Annex G configuration mode. To reset to the default, use the no form of this command.
ttl ttl-value
no ttl
Syntax Description
ttl-value |
Amount of time (in seconds) for which a route from a neighbor is considered valid. Range is from 1 to 2147483647. The default is 1800 (or 30 minutes). |
Command Default
1800 seconds (30 minutes)
Command Modes
Annex G configuration
Command History
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.
Examples
The following example shows a BE with a time-to-live value of 20 seconds.
Router(config)# call-router h323-annexg be20
Router(config-annexg)# ttl 20
Related Commands
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call-router |
Enables the Annex G BE configuration commands. |
show call-router status |
Displays the Annex G BE status. |
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.
type {osp | uni-osp}
no type
Syntax Description
osp |
Enables the Open Settlement Protocol (OSP) server type. |
uni-osp |
Enables authentication of VoIP calls to the Public Switched Telephone Network (PSTN) using a single settlement server. |
Command Default
osp
Command Modes
Settlement configuration
Command History
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12.0(4)XH1 |
This command was introduced on Cisco 2600 series and Cisco 3600 series, and Cisco AS5300. |
12.1(2)T |
The uni-osp keyword was introduced. |
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-osp
Related Commands
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.
type {1 | 2 | 3 | 5}
no type {1 | 2 | 3 | 5}
Syntax Description
Command Default
Type 1
Command Modes
Voice-port configuration
Command History
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|---|---|
11.3(1)T |
This command was introduced on Cisco 3600 series routers. |
11.3(1)MA |
This command was implemented on Cisco MC3810. |
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.
Note E&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.
Examples
The following example selects Type 3 as the interface type for the voice port:
voice-port 1/0/0
type 3
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