HP MSR2000/3000/4000 Router Series Voice Command Reference (V7) Part number: 5998-4018 Software version: CMW710-R0007P02 Document version: 6PW100-20130927
Legal and notice information © Copyright 2013 Hewlett-Packard Development Company, L.P. No part of this documentation may be reproduced or transmitted in any form or by any means without prior written consent of Hewlett-Packard Development Company, L.P. The information contained herein is subject to change without notice.
Contents Voice interface commands ·········································································································································· 1 Analog voice interface commands ·································································································································· 1 area ·······································································································································································
send-busytone time ················································································································································ 37 shutdown ································································································································································ 38 signal ·····································································································································································
display voice entity················································································································································ 79 entity ······································································································································································· 82 ip qos dscp···························································································································································
credentials ···························································································································································· 129 media flow-around ·············································································································································· 130 voice-class sip early-offer forced ························································································································ 131 Call services
Voice interface commands Analog voice interface commands area Use area to specify the standard of busy tones for all the FXO interfaces on the device. Use undo area to restore the default. Syntax area { custom | europe | north-america } undo area Default The European standard is used. Views Voice view Predefined user roles network-admin Parameters custom: Specifies custom busy tones. europe: Specifies the European standard. north-america: Specifies the North American standard.
Parameters index: Assigns a number to a busy tone type, in the range of 0 to 3. The device can record four types of busy tones at most. line-number: Specifies an FXO interface. Usage guidelines The busytone-detect auto command takes effect on only FXO interfaces.
p1: Signal amplitude 1, in the range of 50 to 32767. p2: Signal amplitude 2, in the range of 50 to 32767. p3: Duration of a single tone in milliseconds, in the range of 10 to 1000. p4: Duration error of a single tone in milliseconds, in the range of 0 to 500. p5: Duration of silence in milliseconds, in the range of 10 to 1000. p6: Duration error of silence in milliseconds, in the range of 0 to 500. p7: Absolute difference between p3 and p5 in milliseconds, in the range of 0 to 500.
Examples # Set the number of busy tone detection periods to 3. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] busytone-detect period 3 busytone-hookon delay-timer Use busytone-hookon delay-timer to configure the delay time from when an FXO interface detects a busy tone to when the interface goes on-hook. Use undo busytone-hookon delay-timer to restore the default.
undo calling-name Default No calling name is configured. Views FXS interface view Predefined user roles network-admin Parameters text: Specifies a calling name, a case-sensitive string of 1 to 50 characters. Usage guidelines The calling name can be sent only in the multiple-data-message format. Use this command on the originating device. Examples # Configure the calling name as tony for FXS interface 1/0.
4-wire: Specifies the 4-wire cable type, which provides simplex voice transmission. Every two wires receive and send signals in one direction. Usage guidelines You must configure the same cable type for the E&M interfaces on the originating and terminating devices. Otherwise, only one-way voice communication can be implemented. Examples # Configure the cable type as 2-wire for the E&M interface 1/0.
undo cid receive Default CID receiving is enabled for an FXO interface. Views FXO interface view Predefined user roles network-admin Usage guidelines CID receiving must be enabled for the CID function to work correctly. Examples # Enable CID receiving on FXO interface 1/0.
[Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid ring 0 cid send Use cid send to enable CID sending on an FXS or FXO interface. Use undo cid send to disable CID sending. Syntax cid send undo cid send Default CID sending is enabled on an FXS or FXO interface. Views FXS interface view, FXO interface view Predefined user roles network-admin Usage guidelines CID sending must be enabled for the CID function to work correctly. Examples # Enable CID sending on voice interface 1/0.
simple: Specifies single data message format (SDMF). Usage guidelines The local and remote ends must use the same CID format. The calling name in the CID can only be transmitted in MDMF format. This command takes effect only on the terminating device. Examples # Set the CID format to SDMF on FXS interface 1/0.
Examples # Specify the CID standard as brazil for FXS interface 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid standard-type brazil Related commands cid type cptone Use cptone to specify the call progress tones of a country or region. Use undo cptone to restore the default.
Code Country/region name CZ Czech Republic DK Denmark EG Egypt FI Finland FR France DE Germany GH Ghana GR Greece HK Hong Kong China HU Hungary IS Iceland IN India ID Indonesia IR Iran IE Ireland IEU Ireland (UK style) IL Israel IT Italy JP Japan JO Jordan KE Kenya KR Republic of Korea LB Lebanon LU Luxembourg MO Macau MY Malaysia MX Mexico NP Nepal NL Netherlands NZ New Zealand NG Nigeria NO Norway PK Pakistan PA Panama PH Philippines
Code Country/region name PT Portugal RU Russian Federation SA Saudi Arabia SG Singapore SK Slovakia SI Slovenia ZA South Africa ES Spain SE Sweden CH Switzerland TH Thailand TR Turkey GB United Kingdom US United States UY Uruguay ZW Zimbabwe custom: Customizes call progress tones. busy-tone: Specifies the busy tone. congestion-tone: Specifies the congestion tone. dial-tone: Specifies the dial tone. ringback-tone: Specifies the ringback tone.
time4: Specifies the break time for the second make-to-break ratio in milliseconds. The value range is 0 and 30 to 8191. If time1 is set to 0, this argument must be set to 0. Usage guidelines This command takes effect only for the progress tones on the local device. Examples # Specify the call progress tones of US. system-view [sysname] voice-setup [sysname-voice] cptone country-type us # Customize the call progress tones.
Examples # Set the amplitude of the busy tone to 1200. system-view [sysname] voice-setup [sysname-voice] cptone tone-type busy-tone amplitude 1200 cng-on Use cng-on to enable the comfortable noise generation (CNG) function on a voice interface. Use undo cng-on to disable this function. Syntax cng-on undo cng-on Default The CNG function is enabled.
This command might fail to restore the default settings for some commands for reasons such as command dependencies or system restrictions. Use the display this command in interface view to identify these commands, and then use their undo forms or follow the command reference to individually restore their default settings. If your restoration attempt still fails, follow the error message instructions to resolve the problem. Examples # Restore the default settings for voice interface 1/0.
Syntax delay rising milliseconds undo delay rising Default The delay time is 300 milliseconds. Views E&M interface view Predefined user roles 2: System level Parameters rising milliseconds: Specifies the delay time from when the terminating side detects a seizure signal to when it sends the seizure signal in the delay start mode. The value range is 20 to 2000 milliseconds.
Examples # Set the delay before the originating side sends DTMF tones in immediate start mode to 3000 milliseconds for E&M interface 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] signal immediate [Sysname-subscriber-line1/0] delay send-dtmf 3000 Related commands signal delay send-wink Use delay send-wink to configure a delay from when the terminating side receives a seizure signal to when it sends a wink signal in wink start mode.
Syntax delay start-dial seconds undo delay start-dial Default The dial delay time is 1 second. Views FXS interface view, FXO interface view Predefined user roles network-admin Parameters seconds: Specifies the dial delay time in the range of 0 to 10 seconds. Examples # Set the dial delay time to 5 seconds for voice interface 1/0.
[Sysname-subscriber-line1/0] delay wink-hold 700 Related commands signal delay wink-rising Use delay wink-rising to configure the timeout time for the originating side to wait for a wink signal after sending a seizure signal in wink start mode. Use undo delay wink-rising to restore the default. Syntax delay wink-rising milliseconds undo delay wink-rising Default The timeout time for the originating side to wait for a wink signal after sending a seizure signal is 2000 milliseconds in wink start mode.
Default The description for a voice interface is interface name Interface, for example, Subscriber-line1/0 Interface. Views FXS interface view, FXO interface view, E&M interface view Predefined user roles network-admin Parameters text: Specifies a description, a case-sensitive string of 1 to 80 characters. Examples # Configure the description as pstn for voice interface 1/0.
display voice subscriber-line Use display voice subscriber-line to display information about a voice interface. Syntax display voice subscriber-line line-number Views Any view Predefined user roles network-admin network-operator Parameters line-number: Specifies a voice interface by its number. Examples # Display information about voice interface 1/0.
Field Description • For FXS interfaces: { Idle. { Receiving number. { Ringing. { Listening to ringback tone. { Playing busytone. { Talking. { Releasing. • For FXO interfaces: Call Status { Idle. { Receiving number. { Ringing. { Listening to ringback tone. { Playing busytone { Talking. { Releasing. { Bound and off-hook. { Bound and on-hook. • For E&M interfaces: { Idle. { Sending number. { Ringing. { Listening to ringback tone. { Playing busytone. { Talking.
Examples # Configure the amplitude of DTMF tones as –8.0 dBm. system-view [Sysname] voice-setup [Sysname-voice] dtmf amplitude -8.0 dtmf sensitivity-level Use dtmf sensitivity-level to set the DTMF detection sensitivity level. Use undo dtmf sensitivity-level to restore the default. Syntax dtmf sensitivity-level { high | low | medium [ frequency-tolerance value ] } undo dtmf sensitivity-level Default The DTMF detection sensitivity level is low.
undo dtmf time { interval | persist } Default The duration of DTMF tones and the interval between DTMF tones are both 120 milliseconds. Views Voice view Predefined user roles network-admin Parameters persist: Specifies the duration of DTMF tones. interval: Specifies the interval between DTMF tones. milliseconds: Specifies the time value in the range of 50 to 500 milliseconds. Examples # Set the duration of DTMF tones to 200 milliseconds, and the interval between DTMF tones to 300 milliseconds.
The maximum energy of the input signal in the row frequency group is ROWMAX and the corresponding doubled frequency energy is ROW2nd. The maximum energy in the column frequency group is COLMAX and the corresponding doubled frequency energy is COL2nd. Table 3 Meaning of index numbers Index Meaning Value range Remarks 0 Lower limit of (ROWMAX + COLMAX). The input signal is recognized as a DTMF digit if (ROWMAX + COLMAX) > Value for index 0.
Index Meaning Value range Remarks 9 Lower limit of the DTMF tone duration. The duration of DTMF key tone must be larger than this threshold for the input signal to be recognized as a DTMF digit. 30 to 150 milliseconds, with a default of 30 milliseconds The larger the value, the higher the detection specificity and the lower the detection sensitivity.
Predefined user roles network-admin Parameters convergence-rate value: Sets the convergence rate of comfort noise amplitude, in the range of 0 to 511. The greater the value, the quicker the convergence. max-amplitude value: Sets the maximum amplitude of comfort noise, in the range of 0 to 2048. The greater the value, the greater the noise amplitude. The value 0 indicates that the system performs only nonlinear processing and does not add comfort noise.
Usage guidelines The echo cancellation delay is the time from when a subscriber speaks to when the subscriber hears the echo. Examples # Enable echo cancellation, and configure the echo cancellation delay as 24 milliseconds on voice interface 1/0.
Syntax echo-canceler tail-length milliseconds undo echo-canceler tail-length Default The echo cancellation coverage is 128 milliseconds. Views FXS interface view, FXO interface view, E&M interface view Predefined user roles network-admin Parameters milliseconds: Specifies the echo cancellation coverage in milliseconds. The value range is 32, 48, 64, 80, 96, 112, and 128. Usage guidelines Increasing the echo cancellation coverage can effectively cancel multi-path echoes.
Parameters delay: Specifies the delay off-hook mode. immediate: Specifies the immediate off-hook mode. Examples # Specify the delay off-hook mode for FXO interface 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line 1/0] hookoff-mode delay hookoff-mode delay bind Use hookoff-mode delay bind to bind an FXS interface to an FXO interface. Use undo hookoff-mode delay bind to remove the binding.
Use undo hookoff-time to disable forced on-hook. Syntax hookoff-time time undo hookoff-time Default Forced on-hook is disabled. Views FXO interface view Predefined user roles network-admin Parameters time: Specifies the time from off-hook to forced on-hook, in the range of 60 to 36000 seconds. Usage guidelines In some countries, PBXs do not play busy tones, or the busy tones only last for a short period of time.
Predefined user roles network-admin Parameters country-name: Specifies a country. It can be Australia, Austria, Belgium-Long, Belgium-Short, Brazil, China, Czech-Republic, Denmark, ETSI-Harmanized, Finland, France, German-Swiss, Greece, Hungary, India, Italy, Japan, Korea, Mexico, Netherlands, Norway, Portugal, Slovakia, Spain, Sweden, U.K., US-Loaded-Line, US-Non-Loaded, or US-Special-Service. r550: 550-ohm real impedance. r600: 600-ohm real impedance. r650: 650-ohm real impedance.
Usage guidelines This command takes effect only after the echo-canceler enable command is configured. Examples # Disable the EC nonlinear processing function on voice interface 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line 1/0] undo nlp-on Related commands echo-canceler enable open-trunk Use open-trunk to enable E&M non-signaling mode. Use undo open-trunk to disable E&M non-signaling mode.
system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] open-trunk caller monitor 120 Related commands • private-line • signal passthrough Use passthrough to enable E&M control signals pass-through. Use undo passthrough to disable E&M control signals pass-through. Syntax passthrough undo passthrough Default E&M analog control signals pass-through is disabled.
Predefined user roles network-admin Parameters general: Uses the general compensation mode to reconstruct lost packets. This mode applies to discrete packet loss. specific: Uses the voice gateway-specific compensation mode to reconstruct lost packets. This mode applies to continuous packet loss. Examples # Configure the general compensation mode to reconstruct lost packets.
ring-detect debounce Use ring-detect debounce to configure the debounce time for ring detection. Use undo ring-detect debounce to restore the default. Syntax ring-detect debounce value undo ring-detect debounce Default The debounce time is 10 milliseconds. Views FXO interface view Predefined user roles network-admin Parameters value: Specifies the debounce time for ring detection, in the range of 4 to 15 milliseconds.
Views FXO interface view Predefined user roles network-admin Parameters value: Specifies the frequency value for ring detection, in Hz. The value is in the range 30 to 100 with the step of 10. Examples # Set the frequency value for ring detection on FXO interface 1/0 to 100 Hz. system-view [sysname] subscriber-line 1/0 [sysname-subscriber-line1/0] ring-detect frequency 100 send-busytone enable Use send-busytone enable to enable busy tone sending on an FXO interface.
Syntax send-busytone time seconds undo send-busytone time Default The busy tone duration is 3 seconds. Views FXO interface view Predefined user roles network-admin Parameters time seconds: Specifies the busy tone duration in the range 2 to 15 seconds. Usage guidelines The send-busytone time command takes effect only after you configure the send-busytone enable command. Examples # Enable busy tone sending on FXO interface 1/0, and configure the busy tone duration as 5 seconds.
[Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] shutdown signal Use signal to configure a start mode for an E&M interface. Use undo signal to restore the default. Syntax signal { delay | immediate | wink } undo signal Default An E&M interface uses the immediate start mode. Views E&M interface view Predefined user roles network-admin Parameters delay: Specifies delay start mode. immediate: Specifies immediate start mode. wink: Specifies wink start mode.
Predefined user roles network-admin Parameters threshold: Specifies the silence threshold in the range of 0 to 200. If the amplitude of voice signals from the PBX is smaller than this value, the system regards the voice signals as silence. time-length: Specifies the silence duration for automatic on-hook, in the range of 2 to 7200 seconds. When the silence duration exceeds the specified duration, the FXO interface performs on-hook automatically.
[Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] slic-gain 1 subscriber-line Use subscriber-line to enter voice interface view. Syntax subscriber-line line-number Views System view Predefined user roles network-admin Parameters line-number: Voice interface number. Examples # Enter the view of the voice interface 1/0.
Examples # Configure the maximum interval for dialing the next digit to 5 seconds on voice interface 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer dial-interval 5 timer disconnect-pulse Use timer disconnect-pulse to configure the LCFO signal duration. Use undo timer disconnect-pulse to restore the default. Syntax timer disconnect-pulse value undo timer disconnect-pulse Default The LCFO signal duration is 750 milliseconds.
Views FXS interface view, FXO interface view Predefined user roles network-admin Parameters seconds: Specifies the timeout time between off-hook and dialing the first digit, in the range of 1 to 300 seconds. Usage guidelines If the timer expires before the subscriber dials the first digit, the system prompts the subscriber that the dialing times out. Examples # Configure the timeout time between off-hook and dialing the first digit as 15 seconds.
timer hookoff-interval Use timer hookoff-interval to configure the interval between on-hook and off-hook. Use undo timer hookoff-interval to restore the default. Syntax timer hookoff-interval milliseconds undo timer hookoff-interval Default The interval between on-hook and off-hook is 500 milliseconds. Views FXO interface view Predefined user roles network-admin Parameters milliseconds: Specifies the interval between on-hook and off-hook, in the range of 500 to 4000 milliseconds.
Predefined user roles network-admin Parameters seconds: Specifies the maximum duration for playing ringback tones, in the range of 5 to 120 seconds . Examples # Configure the maximum duration for playing ringback tones as 8 seconds. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer ring-back 8 timer wait-digit Use timer wait-digit to configure the timeout time for the terminating device to wait for the first digit. Use undo timer wait-digit to restore the default.
Syntax transmit gain value undo transmit gain Default The output gain value is 0 dB. Views FXS interface view, FXO interface view, E&M interface view Predefined user roles network-admin Parameters value: Specifies the output gain value in the range of –14.0 to +13.9 dB. Usage guidelines If the power of output voice signals is larger than the power required by the output line, you can use this command to reduce the output gain. Output gain adjustment might lead to call failures.
2: Specifies E&M signal type II. 3: Specifies E&M signal type III. 5: Specifies E&M signal type V. Usage guidelines You must configure the same E&M signal type on the originating and terminating devices. Examples # Configure the signal type as 3 for E&M interface 1/0.
ani-digit Use ani-digit to set the number of dialed digits that the terminating side collects before requesting calling information. Use undo ani-digit to restore the default. Syntax ani-offset number undo ani-offset Default The number of dialed digits that the terminating side collects before requesting calling information is 1. Views R2 CAS view Predefined user roles network-admin Parameters number: Specifies the number of dialed digits, in the range of 1 to 10.
undo answer enable Default The originating side requires the terminating side to send answer signals. Views R2 CAS view Predefined user roles network-admin Usage guidelines If the originating side does not require the terminating side to send answer signals, it directly establishes a call with the terminating side. Otherwise, the originating side establishes a call with the terminating side after receiving answer signals.
[Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] callmode segment cas Use cas to enter R2 CAS view. Use undo cas to exit R2 CAS view and delete the settings in R2 CAS view. Syntax cas ts-set-number undo cas ts-set-number Views E1 interface view, T1 interface view Predefined user roles network-admin Parameters ts-set-number: Specifies a timeslot set by its number in the range of 0 to 30 for an E1 interface, or in the range of 0 to 23 for a T1 interface.
Views R2 CAS view Predefined user roles network-admin Examples # Configure the terminating side to send a clear-back signal when the originating side first disconnects the line. system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] clear-forward-ack enable display voice subscriber-line Use display voice subscriber-line to display information about digital voice interfaces.
TS 1: Idle TS 2: Idle TS 3: Idle TS 4: Idle TS 5: Idle TS 6: Idle TS 7: Idle TS 8: Idle TS 9: Idle TS 10: Idle TS 11: Idle TS 12: Idle TS 13: Idle TS 14: Idle TS 15: Idle TS 17: Idle TS 18: Idle # Display information about voice interface 2/0 generated on a BSV interface. display voice subscriber-line 2/0 Current information : subscriber-line2/0 Type: ISDN Status: Up Call status: TS 0: Idle TS 1: Idle # Display information about a subinterface of voice interface 2/0 generated on a BSV interface.
Field Description • For R2: Call Status { Idle. { Seize. { Seize Ack. { Talking. { Releasing. • For ISDN: { Idle { Call in. { Call out. { Ring. { Ringback tone. { Talking. { Releasing. dl-bits Use dl-bits to configure the ABCD bit pattern for line signals. Use undo dl-bits to restore the default.
Predefined user roles network-admin Parameters answer: Specifies the answer signal. blocking: Specifies the blocking signal. clear-back: Specifies the clear-back signal. clear-forward: Specifies the clear-forward signal. idle: Specifies the idle signal. seizing: Specifies the seizure signal. seizuring-ack: Specifies the seizure acknowledgement signal. release-guard: Specifies the release guard signal. receive: Applies the signaling setting to received line signals.
• MFC—The originating and terminating sides use interregister signaling to transmit and request number information, including the calling number, line information, and billing. In the exchange process, the terminating side sends responses to the originating side. • DTMF—The originating side transmits the called number to the terminating side digit by digit. The terminating side does not send any responses for confirmation.
[Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] final-callednum enable group-b enable Use group-b enable to configure R2 signaling to use Group B signals to complete registers exchange. Use undo group-b enable to configure R2 signaling to not use Group B signals to complete registers exchange. Syntax group-b enable undo group-b enable Default R2 signaling uses Group B signals to complete registers exchange.
Predefined user roles network-admin Parameters line-number: Specifies an E1 or T1 interface by its number. ts-set-number: Specifies a timeslot set by its number. 15: Number for the PRI set created by bundling the timeslots of an E1 interface. 23: Number for the PRI set created by bundling the timeslots of a T1 interface. Examples # Bind a digital voice interface to POTS voice entity 10.
mode Use mode to specify the R2 signaling standard. Use undo mode to restore the default. Syntax mode zone-name [ default-standard ] undo mode Default ITU-T R2 signaling is used. Views R2 CAS view Predefined user roles network-admin Parameters zone-name: Specifies a country or region from the following list: • argentina: Argentina. • australia: Australia. • bengal: Bengal. • brazil: Brazil. • china: China. • custom: Custom. • hongkong: Hong Kong. • india: India. • indonesia: Indonesia.
Examples # Specify the R2 signaling standard of Singapore. system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] mode singapore pcm Use pcm to configure a companding law for PCM. Use undo pcm to restore the default. Syntax pcm { a-law | μ-law } undo pcm Default The companding law for PCM is a-law for E1 interfaces and μ-law for T1 interfaces.
Default Re-answer signal processing is disabled on the originating side. Views R2 CAS view Predefined user roles network-admin Usage guidelines R2 signaling in some countries must support re-answer processing. When the terminating side sends a clear-back signal, the originating side does not release the line, but maintains the call state. If it receives a re-answer signal from the terminating side within a specified time, it continues the call. Otherwise, it disconnects the call upon timeout.
Parameters billingcategory value: Specifies the billing category value in the range of 1 to 16. It configures the KA signal in R2 signaling. The signal provides two types of information for a call connection: billing category (regular, immediate, or toll free) and subscriber level (with or without priority). callcreate-in-groupa value: Specifies the direct call setup signal value in the range of 1 to 16. callingcategory value: Specifies the calling category signal value in the range of 1 to 16.
Usage guidelines The register-value command assigns values to signals requesting responses from the remote end. For example, after you configure the register-value callingcategory command, the terminating side sends the calling category signal with the specified value to the originating side for the calling category. A signal value of 16 disables the signal function. HP recommends that you use the default values.
[Sysname-cas 1/0:0] renew 0011 Related commands mode reverse Use reverse to enable line signal inversion. Use undo reverse to restore the default. Default Line signal inversion is disabled (ABCD takes the value of 0000). Syntax reverse ABCD undo reverse Views R2 CAS view Predefined user roles network-admin Parameters ABCD: Indicates whether corresponding ABCD bits in R2 signaling need inversion. Each argument in this command takes 0 or 1.
Views R2 CAS view Predefined user roles network-admin Parameters max: Selects the timeslot with the greatest number from available timeslots. maxpoll: Selects the timeslot with the greatest number from available timeslots in the first timeslot polling. Subsequent pollings select in descending order timeslots with numbers less than the one selected in the previous polling. Suppose timeslots 31 and 29 are not available. The first polling selects timeslot 30 and the next polling selects timeslot 28.
Examples # Configure the originating side to not require the terminating side to send seizure acknowledgement signals. system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] undo seizure-ack enable send ringbusy enable Use send ringbusy enable to configure the terminating side to send busy tones to the originating side.
Syntax special-character character number undo special-character character number Default No signal code is configured for a special character. Views R2 CAS view Predefined user roles network-admin Parameters character: Specifies a special character, which can be a pound sign (#), asterisk (*), A, B, C, or D. number: Specifies a signal code in the range of 11 to 15.
Usage guidelines Upon creation of a timeslot set on an E1/T1 interface, the system automatically creates a digital voice interface numbered in the form of E1/T1 interface number:timeslot set number. After you create a PRI group with the pri-set command on an E1/T1 interface, the system automatically creates a voice interface numbered E1 interface-number:15 on an E1 interface and T1 interface-number:23 on a T1 interface. Examples # Enter the view of digital voice interface 1/0:15.
• If the line keyword is specified for all interfaces, the clock on the interface with the lowest number is used. If the interface goes down, the clock on the interface with the second lowest number is used. • If the line primary keywords are specified for one interface and the line or internal keyword is specified for all other interfaces, the clock on that one interface is used.
The timer ringbusy command takes effect only after the send ringbusy enable command is configured. Examples # Configure the timeout time for playing ringback tones as 10000 milliseconds. system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] timer ringback 10000 timer dl Use timer dl to set the timeout time of line signals. Use undo timer dl to restore the default.
receive a line signal (for example, clear-back or release guard signal) from the terminating side before the timer expires, it clears the connection. This option applies to the originating side. re-answer time: Specifies the timeout time of re-answer signals, in the range of 100 to 60000 milliseconds. The originating side starts this timer after receiving a clear-back signal from the terminating side.
Examples # Configure the delay for the originating side to send DTMF tones as 800 milliseconds. system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] dtmf enable [Sysname-cas 1/0:0] timer dtmf-delay 800 Related commands dtmf enable timer group-b Use timer group-b to configure the timeout time for Group B signal exchange. Use undo timer group-b to restore the default.
timer register-pulse Use timer register-pulse to configure the duration of register pulse signals. Use undo timer register-pulse to restore the default. Syntax timer register-pulse time undo timer register-pulse Default The duration is 150 milliseconds. Views R2 CAS view Predefined user roles network-admin Parameters time: Specifies the duration of register pulse signals, in the range of 50 to 3000 milliseconds. Examples # Configure the duration of register pulse signals as 300 milliseconds.
timeslots-list: Specifies a timeslot list. Timeslots are numbered 1 through 31 for an E1 interface and 1 through 24 for a T1 interface. The timeslot list can contain one or more individual timeslots separated by commas (for example, 1 or 1, 2, 4), a timeslot range (for example, 5-10), or a combination of the two forms (for example, 1-14, 15, 17-31). signal: Specifies the signaling mode for the timeslot set. r2: R2 signaling.
Examples # Set the trunk direction to bidirectional for timeslot set 0 on interface E1 1/0. system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 1/0] cas 0 [Sysname-cas 1/0:0] trunk-direction timeslots 1-31 dual Related commands select-mode ts Use ts to maintain the circuits of specified timeslots.
Voice entity commands codec Use codec to configure a codec for a voice entity. Use undo codec to delete the configured codec. Syntax codec { g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } [ bytes payload-size ] undo codec Default No codecs are configured. Views POTS entity view, VoIP entity view Predefined user roles network-admin mdc-admin Parameters g711alaw: Specifies the G711 A-Law codec at 64 kbps, which is typically used in Europe.
Table 6 Value range and default of payload-size Codec Value range (in bytes) Default (in bytes) 80 to 240 in multiples of 80 160 g723r53 20 to 120 in multiples of 20 20 g723r63 24 to 144 in multiples of 24 24 g726r16 20 to 220 in multiples of 20 60 g726r24 30 to 210 in multiples of 30 90 g726r32 40 to 200 in multiples of 40 120 g726r40 50 to 200 in multiples of 50 150 10 to 180 in multiples of 10 30 g711alaw g711ulaw g729r8 g729br8 g729r8 Usage guidelines If you configure this com
Examples # Configure the codec as g711alaw for VoIP entity 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] codec g711alaw codec preference Use codec preference to assign a priority to a codec in a codec template. Use the undo codec preference command to delete the assigned priority.
Syntax description string undo description Default No description is configured. Views POTS entity view, VoIP entity view Predefined user roles network-admin mdc-admin Parameters string: Specifies a description, a case-sensitive string of 1 to 80 characters. Examples # Configure the description as room10 for POTS entity 10.
Caller number : 5000 Called number : 1000 Call direction : From packet switch Voice interface index : 0x00000000 Voice entity currently used : 1 Voice entities offered : 1 Table 8 Command output Field Description Call direction • From packet switch—The call is initiated from the IP side. • From circuit switch—The call is initiated from the PSTN side. Voice interface index Index of the voice interface that initiates the call.
Send number: All Max connections: 50 Codec: g723r53; bytes: 80; vad: Disabled Caller permit: 1 Caller group: permit group 1 Substitute called: 9999 Substitute calling: 9999 DTMF relay: Outband-NTE RTP payload-type for NTE: 113 Jitter-buffer mode: Adaptive Jitter-buffer initial delay: 60 ms Jitter-buffer minimum delay: 40 ms Jitter-buffer maximum delay: 200 ms IP media DSCP: ef IP signaling DSCP: ef Register number: Enabled Call-forwarding no-reply number: 5555 Call-forwarding on-busy number: 6666 Call-forwa
Voice class codec: 1 Keepalive up-interval: 50 s Keepalive down-interval: 30 s Keepalive retry: 3 Table 9 Command output Field VoIP entity-number Description Voice entity type and number. The voice entity type can be VoIP or POTS. Match template Number template of the voice entity. Voice line Voice interface bound to the voice entity. line Voice interface bound to the voice entity. Number sending mode: Send number • All—Sends all digits of a called number.
Field Description Call-forwarding unavailable number Destination number to which incoming calls will be forwarded when the voice interface is shut down by executing the shutdown command. Call-forwarding unconditional number Destination number to which incoming calls will be forwarded, whether or not the voice interface is available. Keepalive up-interval Interval for the local end to send OPTIONS messages before marking the voice entity unavailable.
[Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots ip qos dscp Use ip qos dscp to set the DSCP value of IP packets carrying streaming media or signaling. Use undo ip qos dscp to restore the default. Syntax ip qos dscp { dscp-value | dscp-value-set } { media | signaling } undo ip qos dscp { dscp-value | dscp-value-set } { media | signaling } Default The DSCP value is ef (101110).
Keyword DSCP value in binary DSCP value in decimal cs1 001000 8 cs2 010000 16 cs3 011000 24 cs4 100000 32 cs5 101000 40 cs6 110000 48 cs7 111000 56 ef 101110 46 Examples # Set the DSCP value of IP packets carrying signaling to af41. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] ip qos dscp af41 signaling line Use line to bind a voice interface to a POTS entity.
[Sysname-voice-dial-entity10] line 1/0 match-template Use match-template to configure the number template for a voice entity. Use undo match-template to remove the configuration. Syntax match-template match-string undo match-template Default No number template is configured for a voice entity.
Character Description Dot (.) Wildcard, which can match any valid digit. For example, 555…. can match any 7-digit number beginning with 555. Exclamation point (!) Indicates the sub-expression before it appears once or does not appear. For example, 56!1234 can match 51234 and 561234. Plus sign (+) Indicates the sub-expression before it appears one or more times. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on.
outband nte Use outband nte to configure out-of-band NTE DTMF transmission. Use undo outband to restore the default. Syntax outband nte undo outband Default Inband DTMF transmission is adopted. Views POTS entity view, VoIP entity view, Predefined user roles network-admin mdc-admin Usage guidelines HP recommends that you configure the outband nte command and the same payload type value on the originating and terminating devices. Otherwise, DTMF tones might fail to be transmitted.
Predefined user roles network-admin mdc-admin Parameters value: Specifies the value of the payload type field in RTP packets, in the range of 96 to 127. Usage guidelines Do not set the payload type field to 98, which has been used to identify nonstandard T38 fax packets. When the device is connected to a device from another vendor, you cannot set the payload type field to any value forbidden by that device. Otherwise, NTE negotiation might fail.
[Sysname-voice-dial-entity10] shutdown vad-on Use vad-on to enable VAD. Use undo vad-on to disable VAD. Syntax vad-on [ g723r53 | g723r63 | g729a | g729r8 ] * undo vad-on [ g723r53 | g723r63 | g729a | g729r8 ] * Default VAD is disabled. Views POTS entity view, VoIP entity view Predefined user roles network-admin mdc-admin Parameters g723r53: Specifies the G723.1 Annex A codec at 5.3 kbps. g723r63: Specifies the G723.1 Annex A codec at 6.3 kbps. g729a: Specifies the G729 Annex A codec at 8 kbps.
undo voice class codec tag Default No codec templates exist. Views Voice view Predefined user roles network-admin mdc-admin Parameters tag: Specifies the number of the codec template, in the range of 1 to 2147483647. Usage guidelines The device supports a maximum of 16 codec templates. Examples # Create codec template 1.
Only one codec template can be bound to a voice entity. If you configure the voice-class codec command multiple times, the most recent configuration takes effect. Examples # Bind codec template 1 to VoIP entity 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] voice-class codec 1 Related commands • codec preference • voice class codec voice-setup Use voice-setup to enter voice view and enable voice services.
Dial program commands caller-group Use caller-group to configure a voice entity to permit or deny the calling numbers in a subscriber group. Use undo caller-group to remove the configuration. Syntax caller-group { deny | permit } group-id undo caller-group { { deny | permit } group-id | all } Default A voice entity permits all calling numbers. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters deny: Denies the calling numbers in the subscriber group.
Default A voice entity permits all calling numbers. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters all: Removes all the configured calling numbers. calling-string: Specifies a string of 1 to 31 characters in the format of { [ + ] string [ $ ] }| $. The voice entity uses the string to match calling numbers. The following describes the symbols in the format: • Plus sign (+): If the plus sign (+) is at the beginning of the string, the string indicates an E.
Character Description Brackets ([ ]) Indicates a range. For example, [1-36A] matches 1, 2, 3, 6, or A. Parentheses (( )) Indicates a string of characters. For example, (123) indicates the character string 123. It is usually used together with signs such as !, %, and +. For example, 408(12)+ can match the character string 40812 or 408121212, but not 408 (that is, the string 12 must appear at least one time). If embedded, brackets([ ]) and parentheses (( )) must be presented in the form of "( [ ] )".
[Sysname-voice-dial] subscriber-group 10 [Sysname-voice-dial-group10] description international dial-prefix Use dial-prefix to configure a dial prefix for a POTS entity. Use undo dial-prefix to remove the dial prefix of a POTS entity. Syntax dial-prefix string undo dial-prefix Default No dial prefix is configured for a voice entity.
dial-program Use dial-program to enter dial program view. Use undo dial-program to remove all the settings from dial program view. Syntax dial-program undo dial-program Views Voice view Predefined user roles network-admin Examples # Enter dial program view. system-view [Sysname] voice-setup [Sysname-voice] dial-program dot-match Use dot-match to configure a dot match rule. Use undo dot-match to restore the default.
Examples # Set the dot match rule to right-left for number substitution rule list 20. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 20 [Sysname-voice-dial-substitute20] dot-match right-left Related commands rule first-rule Use first-rule to configure the preferred number substitution rule. Use undo first-rule to remove the preferred number substitution rule.
match-template Use match-template to configure a calling number match template for a subscriber group. Use undo match-template to delete a calling number match template or all calling number match templates from a subscriber group. Syntax match-template match-string undo match-template { match-string | all } Default No calling number match template is configured for a subscriber group.
Character Description Plus sign (+) Indicates the sub-expression before it appears one or more times. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on. Percent sign (%) Indicates the sub-expression before it appears multiple times or does not appear. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on. Hyphen (-) the sign is similar to that of the wildcard dot (.). These signs must follow a valid digit or digit string.
Predefined user roles network-admin Parameters max-number: Specifies the maximum number of total calls allowed by a voice entity. Examples # Configure the maximum number of total calls as 100. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] max-conn 100 number-match Use number-match to configure a global number match mode. Use undo number-match to restore the default number match mode.
Use undo number-substitute to remove a number substitution rule list or all number substitution rule lists. Syntax number-substitute list-number undo number-substitute { list-number | all } Default No number substitution rule list is configured. Views Voice dial program view Predefined user roles network-admin Parameters list-number: Specifies an ID for the number substitution rule list, in the range of 1 to 2147483647. all: Specifies all number substitution rule lists.
Examples # Set the priority to 5 for POTS entity 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] priority 5 private-line Use private-line to configure the private line auto ring-down (PLAR) function. Use undo private-line to disable the PLAR function. Syntax private-line string undo private-line Default The PLAR function is disabled.
Default No number substitution rule is configured. Views Number-substitute view Predefined user roles network-admin Parameters all: Deletes all number substitution rules. id: Specifies an ID for the substitution rule, in the range of 0 to 31. input-template: Configures an input template of 1 to 31 characters in the format of [ ^ ] [ + ] string [ $ ]. The following describes the signs in the format: • Caret (^): Indicates the match begins with the first character of the string.
Table 16 Input number types Number type Description abbreviated Abbreviated number. any Any number. international International number. national National number that is not in the local network. network Specific service network number. reserved Reserved number. subscriber Local network number. unknown Number of an unknown type. output-template-type: Specifies an output number type. Table 17 describes the values for this argument.
Table 19 Output numbering plans Numbering plan Description data Data numbering plan. isdn ISDN telephone numbering plan. national National numbering plan. private Private numbering plan. reserved Reserved numbering plan. telex Telex numbering plan. unknown Unknown numbering plan. Usage guidelines The following describes the functions of dots in the input-template and output-template arguments: • Dots in the output-template argument are invalid if the dot match rule is end-only.
Examples # Configure a number substitution rule for number substitution rule list 1: the input template is ^..01...$, and the output template is ...1. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 1 [Sysname-voice-dial-substitute1] rule 0 ^..01...$ ...1 Related commands • dot-match • first-rule • substitute (Voice dial-program view) • substitute (Voice entity view) send-number Use send-number to configure the number sending mode.
Related commands match-template subscriber-group Use subscriber-group to create a subscriber group and enter subscriber group view. Use undo subscriber-group to delete a subscriber group or all subscriber groups. Syntax subscriber-group group-id undo subscriber-group { group-id | all } Default No subscriber group is created. Views Voice dial program view Predefined user roles network-admin Parameters group-id: Specifies a subscriber group ID in the range of 1 to 2147483647.
Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters called: Applies the number substitution rule to called numbers. calling: Applies the number substitution rule to calling numbers. list-number: Specifies a number substitution rule list by its number in the range of 1 to 2147483647. Examples # Apply number substitution rule list 6 to called numbers on POTS entity 10.
list-number: Specifies a number substitution rule list by its number in the range of 1 to 2147483647. Usage guidelines You can apply up to 32 number substitution rule lists. Examples # Apply number substitution rule list 5 to the called numbers of incoming calls. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] substitute incoming-call called 5 # Apply number substitution rule lists 5, 6, and 8 to the called numbers of outgoing calls.
SIP commands address sip Use address sip to configure a call destination IP address for a VoIP voice entity. Use undo address sip to remove the call destination IP address for a VoIP voice entity. Syntax address sip { dns domain-name port port-number | ip ip-address [ port port-number ]| proxy } undo address sip { dns | ip | proxy } Default No call destination IP address is configured for a VoIP voice entity.
Syntax asserted-id { pai | ppi } undo asserted-id Default SIP messages do not carry the P-Asserted-Identity or P-Preferred-Identity header field. Views SIP view Predefined user roles network-admin Parameters pai: Adds the P-Asserted-Identity header field into SIP messages. ppi: Adds the P-Preferred-Identity header field into SIP messages. Examples # Add the P-Asserted-Identity header field into SIP messages.
Destination IP address : 192.168.4.69: 5060 Media stream Media status : None Stream type : Voice Negotiated codec : g729r8 Codec payload type : 18 Codec payload size : 30 Codec transparent : Disabled Media mode : Flow-through Negotiated DTMF-relay : Inband-voice Source IP address : 192.168.4.98: 16334 Destination IP address : 192.168.4.69: 16342 Number of SIP UAC calls: 1 Table 20 Command output Field Description Call status, including: Call status • • • • Originating. Answering.
sip-pstn: Displays SIP status-to-PSTN cause mappings. Examples # Display PSTN cause-to-SIP status mappings.
Field Description SIP status code. SIP-Status If the configured SIP status code is different from the default, it is highlighted with an asterisk. Default Default SIP status code. # Display SIP status-to-PSTN cause mappings.
37 606 58 58 Table 22 Command output Field Description SIP-Status SIP status code. PSTN cause code. PSTN-Cause If the configured PSTN cause code is different from the default, it is highlighted with an asterisk. Default Default PSTN cause code. display voice sip register-status Use display voice sip register-status to display SIP UA registration status information.
Field Description State of the phone number, including: Status • • • • • • Offline. Online. Login. Logout. DNS-in—DNS query is being performed before the number is registered. DNS-out—DNS query is being performed before the number is deregistered. min-se Use min-se to set the maximum and minimum session expiration timers. Use undo min-se to restore the default.
outband sip Use outband sip to enable out-of-band DTMF. Use undo outband sip to restore the default. Syntax outband sip undo outband Default Inband DTMF is enabled. Views POTS entity view, VoIP entity view Predefined user roles network-admin Usage guidelines If you use out-of-band DTMF, configure the outband sip command on both the calling and called devices. Examples # Enable out-of-band DTMF for VoIP entity 10.
[Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] privacy proxy Use proxy to configure proxy server information for a SIP UA. Use undo proxy to remove the proxy server information for a SIP UA. Syntax proxy { dns domain-name port port-number | ip ip-address [ port port-number ] } undo proxy { dns | ip } Default No proxy server information is configured for a SIP UA.
Syntax register-number undo register-number Default After you complete the SIP registration configuration, a POTS entity registers the phone number with the registrar. Views POTS entity view Predefined user roles network-admin Usage guidelines A registrar cannot store multiple entries for one phone number. Therefore, if multiple POTS entities on a device have the same phone number, only one POTS entity can register that number with the registrar.
Parameters registrar-index: Specifies an index for the registrar, in the range of 1 to 6. ip ip-address: Specifies the registrar by its IP address. port port-number: Specifies the port number of the registrar, in the range of 1 to 65535. If the ip keyword is specified, the default port number is 5060. If the dns keyword is specified, the port number must be configured. expires seconds: Sets the registration expiry time in the range of 60 to 65535 seconds.
Predefined user roles network-admin Examples # Add the Remote-Party-ID header field into INVITE requests. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] remote-party-id session refresh Use session refresh to enable SIP session refresh. Use undo session refresh to disable SIP session refresh. Syntax session refresh undo session refresh Default A UAC does not perform SIP session refresh. A UAS supports SIP session refresh.
Default Table 24 shows the default PSTN cause-to-SIP status mappings. Table 24 Default PSTN cause-to-SIP status mappings PSTN cause code PSTN cause description SIP status code SIP status description 1 Unallocated (unassigned) number! 404 Not Found. 2 No route to specified transit network! 404 Not Found. 3 No route to destination! 404 Not Found. 16 Normal clearing! N/A BYE or CANCEL. 17 User busy! 486 Busy here. 18 No user responding! 408 Request Timeout.
PSTN cause code PSTN cause description SIP status code SIP status description 63 Service or option not available, unspecified! 500 Server internal error. 65 Bearer capability not implemented! 488 Not Acceptable Here. 70 Only restricted digital information bearer capability is available! 488 Not Acceptable Here. 79 Service or option not implemented, unspecified! 501 Not implemented. 87 User not member of Closed User Group (CUG)! 403 Forbidden.
Table 25 Default SIP status-to-PSTN cause mappings SIP status code SIP status description PSTN cause code PSTN cause description 400 Bad Request. 41 Temporary failure! 401 Unauthorized. 21 Call rejected! 402 Payment required. 21 Call rejected! 403 Forbidden. 21 Call rejected! 404 Not found. 1 Unallocated (unassigned) number! 405 Method not allowed. 63 Service or option not available, unspecified! 406 Not acceptable.
SIP status code SIP status description PSTN cause code PSTN cause description 503 Service unavailable. 41 Temporary failure! 504 Server time-out. 102 Recovery on timer expiry! 505 Version Not Supported. 127 Interworking, unspecified! 513 Message Too Large. 127 Interworking, unspecified! 600 Busy everywhere. 17 User busy! 603 Decline. 21 Call rejected! 604 Does not exist anywhere. 1 Unallocated (unassigned) number! 606 Not acceptable.
system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] user Use user to configure SIP credentials. Use undo user to restore the default. Syntax user username password { cipher | simple } password [ realm realm ] undo user [ username password { cipher | simple } password [ realm realm ] ] Default No SIP credentials are configured.
Examples # Configure global SIP credentials that include username abcd and plaintext password 1234. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] user abcd password simple 1234 # Configure SIP credentials that include username abcd, plaintext password 1234, and realm abc for POTS entity 100.
SIP trunk commands allow-connections sip to sip Use allow-connections sip to sip to enable SIP-to-SIP calling. Use undo allow-connections sip to sip to disable SIP-to-SIP calling. Syntax allow-connections sip to sip undo allow-connections sip to sip Default SIP-to-SIP calling is disabled. Views Voice view Predefined user roles network-admin Usage guidelines After you enable SIP-to-SIP calling, the device works as a SIP trunk device. HP does not recommend using a SIP trunk device as a SIP UA.
Predefined user roles network-admin Usage guidelines If the SIP trunk device does not support any codecs on the calling and called parties, you can enable codec transparent transmission. The SIP trunk device transparently forwards codec capability sets between the two parties without intervening codec negotiation. Examples # Enable codec transparent transmission for VoIP entity 1.
command, and it uses the realm value in 401/407 responses from the registrars to identify the matching credentials. You can configure up to 12 realms for a phone number, and up to 128 SIP trunk accounts on the device. Examples # Configure a SIP trunk account for phone number 1000 that uses username 1000 and password 1000, for realm server1, uses username 2000 and password 2000 for realm server2 , and uses username 3000 and password 3000 for realm server3.
[Sysname-voice-dial-entity1] media flow-around voice-class sip early-offer forced Use voice-class sip early-offer forced to enable delayed offer to early offer (DO-EO) conversion. Use undo voice-class sip early-offer forced to restore the default. Syntax voice-class sip early-offer forced undo voice-class sip early-offer forced Default DO-EO conversion is disabled.
Call services commands call-forwarding Use call-forwarding to configure call forwarding. Use undo call-forwarding to restore the default. Syntax call-forwarding { no-reply | on-busy | unavailable | unconditional } number number undo call-forwarding { no-reply | on-busy | unavailable | unconditional } Default Call forwarding is disabled. Views POTS voice entity view Predefined user roles network-admin mdc-admin Parameters no-reply: Enables call forwarding no reply. on-busy: Enables call forwarding busy.
[Sysname-voice-dial-entity10] call-forwarding no-reply number 12345678 # Enable call forwarding busy and set the forwarded-to number to 12345678. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] call-forwarding on-busy number 12345678 # Enable call forwarding unavailable and set the forwarded-to number to 12345678.
Examples # Configure the call hold mode as sendonly.
Support and other resources Contacting HP For worldwide technical support information, see the HP support website: http://www.hp.
Conventions This section describes the conventions used in this documentation set. Command conventions Convention Description Boldface Bold text represents commands and keywords that you enter literally as shown. Italic Italic text represents arguments that you replace with actual values. [] Square brackets enclose syntax choices (keywords or arguments) that are optional. { x | y | ... } Braces enclose a set of required syntax choices separated by vertical bars, from which you select one.
Network topology icons Represents a generic network device, such as a router, switch, or firewall. Represents a routing-capable device, such as a router or Layer 3 switch. Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that supports Layer 2 forwarding and other Layer 2 features. Represents an access controller, a unified wired-WLAN module, or the switching engine on a unified wired-WLAN switch. Represents an access point.
Index ABCDEFGHILMNOPRSTUVW A D address sip,110 default,14 allow-connections sip to sip,128 delay hold,15 ani,47 delay rising,15 ani-digit,48 delay send-dtmf,16 answer enable,48 delay send-wink,17 area,1 delay start-dial,17 asserted-id,110 delay wink-hold,18 B delay wink-rising,19 description,94 busytone-detect auto,1 description,19 busytone-detect custom,2 description,77 busytone-detect period,3 dial-prefix,95 busytone-hookon delay-timer,4 dial-program,96 C disconnect lcfo,20 cab
receive gain,35 final-callednum enable,55 first-rule,97 register-number,118 G register-value,60 registrar,119 group-b enable,56 remote-party-id,120 H renew,62 hookoff-mode,29 reverse,63 hookoff-mode delay bind,30 ring-detect debounce,36 hookoff-time,30 ring-detect frequency,36 I rtp payload-type nte,87 rule,102 impedance,31 ip qos dscp,83 S L seizure-ack enable,64 select-mode,63 line,84 send ringbusy enable,65 line,56 send-busytone enable,37 M send-busytone time,37 match-template,
timer hookflash-detect,43 user,126 timer hookoff-interval,44 V timer register-pulse,72 vad-on,89 timer ring-back,44 voice class codec,89 timer wait-digit,45 voice-class codec,90 timeslot-set,72 voice-class sip early-offer forced,131 transmit gain,45 voice-setup,91 trunk-direction,73 ts,74 W type,46 Websites,135 U 140
