HP MSR Router Series Voice Command Reference(V5) Part number: 5998-2047 Software version: CMW520-R2511 Document version: 6PW103-20140128
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Contents Voice entity configuration commands ························································································································ 1 area-id ······································································································································································· 1 call-history ·························································································································································
Voice subscriber line configuration commands ······································································································· 55 Analog voice subscriber line configuration commands ····························································································· 55 area ········································································································································································ 55 audio-input-gain ·················
silence-th-span ························································································································································ 96 slic-gain ·································································································································································· 97 subscriber-line ····················································································································································
timer register-complete group-b ························································································································· 143 timer ring ······························································································································································ 143 timeslot-set ···························································································································································· 144 trunk-
proxy ····································································································································································· 195 reason-mapping pstn··········································································································································· 196 reason-mapping sip············································································································································· 198 register-
assign···································································································································································· 236 account enable ···················································································································································· 237 bind sip-trunk account ········································································································································· 238 code
call-waiting enable ·············································································································································· 274 call-waiting priority·············································································································································· 275 conference enable ··············································································································································· 276 dialin-rest
select-rule operation-order ·································································································································· 319 set-media ······························································································································································ 320 timeout ·································································································································································· 321 u
Voice entity configuration commands area-id Use area-id to configure the area ID of the voice GW. Use undo area-id to remove the specified area ID. Syntax area-id string undo area-id Default No area ID is configured. Views VoIP voice entity view, VoFR entity view Default command level 2: System level Parameters string: Area ID, a string of 1 to 31 characters, which consists of digits 0 through 9 and the pound sign (#).
Default The maximum number of call history records that can be stored is 50. Views Voice view Default command level 2: System level Parameters number: Maximum number of call history records that can be stored, in the range of 0 to 200. Examples # Configure the maximum number of call history records that can be stored as 100.
g723r63: G.723.1 Annex A codec, requiring a bandwidth of 6.3 kbps. g726r16: G.726 Annex A codec. It uses the adaptive differential pulse code modulation (ADPCM) technology, requiring a bandwidth of 16 kbps. g726r24: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 24 kbps. g726r32: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 32 kbps. g726r40: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 40 kbps. g729a: G.729 Annex A codec (a simplified version of G.
Table 2 G.711 algorithm (A-law and μ-law) Packet assembly interval Bytes coded in a time unit Packet length (IP) (bytes) Network bandwidth (IP) Packet length (IP+PPP) (bytes) Network bandwidt h (IP+PPP) Coding latency 10 ms 80 120 96 kbps 126 100.8 kbps 10 ms 20 ms 160 200 80 kbps 206 82.4 kbps 20 ms 30 ms 240 280 74.7 kbps 286 76.3 kbps 30 ms G.711 algorithm (A-law and μ-law): media stream bandwidth 64 kbps, minimum packet assembly interval 10 ms. Table 3 G.
Packet assembly interval Bytes coded in a time unit Packet length (IP) (bytes) Network bandwidth (IP) Packet length (IP+PPP) (bytes) Network bandwidth (IP+PPP) Coding latency 20 ms 40 80 32 kbps 86 34.4 kbps 20 ms 30 ms 60 100 26.7 kbps 106 28.3 kbps 30 ms 40 ms 80 120 24 kbps 126 25.2 kbps 40 ms 50 ms 100 140 22.4 kbps 146 22.1 kbps 50 ms 60 ms 120 160 21.3 kbps 166 11.4 kbps 60 ms 70 ms 140 180 20.6 kbps 186 21.
Table 8 G.726 r40 algorithm Packet assembly interval Bytes coded in a time unit Packet length (IP) (bytes) Network bandwidt h (IP) Packet length (IP+PPP) (bytes) Network bandwidth (IP+PPP) Coding latency 10 ms 50 90 72 kbps 96 76.8 kbps 10 ms 20 ms 100 140 56 kbps 146 58.4 kbps 20 ms 30 ms 150 190 50.7 kbps 196 52.3 kbps 30 ms 40 ms 200 240 48 kbps 246 49.2 kbps 40 ms G.726 r40 algorithm: media stream bandwidth 40 kbps, minimum packet assembly interval 10 ms. Table 9 G.
Network bandwidth = Bandwidth of the media stream × packet length/bytes coded in a time unit. Because IPHC compression is affected significantly by network stability, it cannot achieve high efficiency unless the line is of high quality, the network is very stable, and packet loss does not occur or seldom occurs. When the network is unstable, IPHC efficiency drops drastically. With best IPHC performance, the IP (RTP) header can be compressed to 2 bytes.
Default The codec with the first priority is g729r8, that with the second priority is g711alaw, that with the third priority is g711ulaw, and that with the fourth priority is g723r53. Views Voice dial program view Default command level 2: System level Parameters 1st-level: Specifies a codec with the first priority. 2nd-level: Specifies a codec with the second priority. 3rd-level: Specifies a codec with the third priority.
default entity payload-size Use default entity payload-size to configure the default packetization period for a codec. Use undo default entity payload-size to restore the default.
Examples # Set the packetization period for G.711 codec to 30 milliseconds. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] default entity payload-size g711 30 default entity vad-on Use default entity vad-on to globally configure voice activity detection (VAD) as the default value. Use undo default entity vad-on to restore the fixed value (that is, disabling VAD) to be the default value.
Default No description is configured for the voice entity. Views POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view Default command level 2: System level Parameters string: Specifies a description of 1 to 80 characters for the voice entity. Usage guidelines Use the description command to add a description to a voice entity, which has no effect on the performance of the voice entity interface. View this description with the display command.
dial-program Use dial-program to enter the voice dial program view. Syntax dial-program Views Voice view Default command level 2: System level Examples # Enter the dial program view system-view [Sysname] voice-setup [Sysname-voice] dial-program display voice call-info Use display voice call-info to display the contents in the call information table.
ViIfIndex : 0x002C0060 Module ID : LGS CMC # End # Display the detailed information of the call information table at a certain point of time.
Parameters ccb: Displays the call control block of the call management center (CMC) module. statistic: Displays statistics related to the CMC module. all: Displays all statistics related to the CMC module. em: Displays EM module information related to the CMC module. h323: Displays H.323 module information related to the CMC module. iva: Displays IVA module information related to the CMC module. lgs: Displays relevant LGS module information related to the CMC module.
} OUTGOING CALLLEG NUMBER : 1 OUTGOING LEG[0] { Spl Protocol : LGS LocalRef : 0x0003 IfIndex : 2884064 IpAddress : 0.0.0.0 IpPort : 0 LegState : OUT_STATE_ACTIVE ConnectState : CONN_STATE_ACTIVE } # End.
Field Description CallerAddr Caller number of the call. CallerAddrSubst Caller number after substitution. CallInfoTabIndex Call information index of the call. Call Leg Number Number of call legs of the call. Active Service Number of services involved in the call. Spl Protocol Type of protocol used in the call leg. LocalRef Local call identifier of the call leg. IfIndex Voice interface index connected to the call leg. IpAddress IP address connected to the call leg.
begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters. Examples # Display the current default values and the system-default values.
Field Description fax train-mode Fax training mode. fax cng-switch CNG fax switch. compression 1st-level Voice coding mode of the first preference. compression 2nd-level Voice coding mode of the second preference. compression 3rd-level Voice coding mode of the third preference. compression 4th-level Voice coding mode of the fourth preference. vad-on Voice entity VAD. payload-size g711 Voice entity packet assembly interval (G.711) payload-size g723 Voice entity packet assembly interval (G.
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
h225: Displays statistics about H.225 messages. h245: Displays statistics about H.245 messages. ras: Displays statistics about RAS messages. socket: Displays statistics about socket messages. timer: Displays timeout statistics. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow.
Recv_FacilityOLCRequest : 0 Recv_FacilityOLCAck : 0 Recv_FacilityOLCReject : 0 Recv_FacilityMSDRequest : 0 Recv_FacilityMSDAck : 0 Recv_FacilityMSDReject : 0 Recv_FacilityCLCRequest : 0 Recv_FacilityCLCAck : 0 Recv_Unknown : 0 } Table 15 Command output Field Description Setup Statistics of Setup messages. CallProceeding Statistics of CallProceeding messages. Alerting Statistics of Alerting messages. Connect Statistics of Connect messages.
Views Any view Default command level 2: System level Parameters all: Displays all the statistic information related to the IVA module. call: Displays the calling statistics in the IVA module. cmc: Displays all the interaction statistics between the IVA and the CMC module. error: Displays all the error statistics of the IVA module. isdn: Displays the interaction statistics between IVA module and ISDN. proc: Displays the statistic information of process call in the IVA module.
Field Description IVA_ISDN_PASSIVE_CALL_FAILED Statistics of failed calls when IVA serves as the called. display voice jitter-buffer Use display voice jitter-buffer to view the jitter buffer statistics of the last call. Syntax display voice jitter-buffer subscriber-line line-number [ | { begin | exclude | include } regular-expression ] Views Any view Default command level 2: System level Parameters subscriber-line line-number: Specifies the voice subscriber line number.
Delay Adjust = 0 JB Delay Total = 0 ms DSP Delay Total = 0 ms JB Full = 99 DSP Full = 0 RTP Unexpected = 0 JB Errors = 0 Table 17 Command output Field Description Payload Size Packet assembly interval. JB Discard Number of times the jitter buffer has discarded packets. Delay Adjust Number of times the jitter buffer has performed delay adjust. JB Delay Dec Declined delay time delivered to the DSP in total. DSP Delay Dec Delay time actually declined by the DSP in total.
When multiple calls are in progress, the call records are displayed in chronological order. Examples # Display the statistics of all active calls.
CallOrigin:Originate ChargedUnits:0 CallInfoType:Speech ByteReceived:115068030 ByteTransmitted:115067484 PacketReceived:2739715 PacketTransmitted:2739702 PSTN Info: ConnectionId:0x0013 CallId:1 TxDuration:82191625 ms VoiceTxDuration:82191060 ms FaxTxDuration:0 ms ImgPages:0 CodecType:G729r8 CallingNumber:200 CalledNumber:100 SubstCallingNumber:200 SubstCalledNumber:100 End Table 18 Command output Field Description SetupTime Length of the time from the system starts up to the start time of the call, in m
Field Description ConnectionId Connection ID, which is used to identify a call. CallId Identification number of the calling side. RemoteSignallingIPAd dress IP address of the remote signaling. RemoteSignallingPort Port number of the remote signaling. RemoteMediaIPAddr IP address of the remote media. RemoteMediaPort Port number of the remote media. SessionProtocol Session protocol type. Only the SIPv2 protocol is supported. CallingNumber Calling number before the substitution.
Parameters all: Displays the history records of all calls that have ended. If this keyword is provided, the number of call history records can be displayed depends on the maximum number of call history records that can be stored, which is specified with the call-history command. last index: Displays the history record of the specified call that has ended. The value range for the index argument is 1 to 100. |: Filters command output by specifying a regular expression.
Call-History Info: Index:2 SetupTime:155448 ms PhoneNumber:6001 EntityIndex:6000 IfIndex:0x0 ConnectTime:168011 ms TerminateTime:171131 ms CallOrigin:Answer ChargedUnits:0 CallInfoType:Speech ByteReceived:21798 ByteTransmited:18816 PacketReceived:519 PacketTransmited:448 VOIP Info: ConnectionId:0x0000 CallId:0 RemoteSignallingIPAddress: 100.1.1.223 RemoteSignallingPort:5060 RemoteMediaIPAddress:100.1.1.
Field Description CallInfoType Information type for this call, Speech or Fax. CallId Identification number of the calling side. RemoteSignallingIPAddress Remote signaling IP address. RemoteSignallingPort Remote signaling port number. RemoteMediaIPAddr Remote media IP address. RemoteMediaPort Remote media port number. SessionProtocol Session protocol type. Only SIPv2 is supported. CallingNumber Calling number before the substitution. CalledNumber Called number before the substitution.
Default command level 1: Monitor level Parameters all: Displays the call statistics of all voice entities. mark entity-index: Displays the call statistics of the specified entity. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression.
Field Description ConnectTime Accumulated connect time to the peer since the system started up, in milliseconds. LastSetupTime Setup time of the last call, in milliseconds. distinguish-localtalk Use distinguish-localtalk to enable the local call identification function. Use undo distinguish-localtalk to disable this function. Syntax distinguish-localtalk undo distinguish-localtalk Default The local call identification function is disabled.
Default command level 2: System level Parameters dscp-value: DSCP value in the range of 0 to 63 or the keyword af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, and ef. Examples # Set the DSCP value in the ToS field of the IP packets that carry the RTP stream of VoIP voice entity to af41.
system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots Related commands line fast-connect Use fast-connect to enable H.323 fast connection. Use undo fast-connect to disable H.323 fast connection. Syntax fast-connect undo fast-connect Default Fast connection is disabled.
fxo-monitoring enable Use fxo-monitoring enable to enable online monitoring for all FXO ports. Use undo fxo-monitoring enable to disable online monitoring for all FXO ports. Syntax fxo-monitoring enable undo fxo-monitoring enable Default Online monitoring is enabled for all FXO ports. Views Voice view Default command level 2: System level Usage guidelines This feature monitors the state of all FXO ports in real time. Examples # Disable online monitoring for all FXO ports.
[Sysname-voice-dial-entity100] jitter-buffer mode adaptive jitter-buffer delay Use jitter-buffer delay to set the operating parameters for the jitter buffer that operates in the adaptive mode. Use undo jitter-buffer delay to restore the default. Syntax jitter-buffer delay { initial milliseconds | maximum milliseconds } undo jitter-buffer delay { initial | maximum } Default The initial buffering time is 30 milliseconds and the maximum buffering time is 160 milliseconds.
Parameters interval seconds: Sets the time interval for sending the options keepalive packets, in seconds. The value range is 5 to 65535, and the default is 60. Usage guidelines The keepalive function takes effect only when the destination address of the VoIP voice entity is a DNS domain name or an IP address. Examples # Set the keepalive interval for the VoIP voice entity 203 to 180 seconds.
Syntax match-template match-string undo match-template Default No number template is bound to the local voice subscriber line in POTS view, no number template is configured for the terminating side when the POTS voice entity serves as a trunk, and no number template is configured for the voice entity in VoIP, VoFR, or IVR entity view. Views POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view Default command level 2: System level Parameters match-string: Number template.
Character Meaning Brackets ([ ]) Select one character from the group. For example, [1-36] can match only one character among 1, 2, 3, and 6. Parentheses (( )) A group of characters. For example, (123) means a string "123". It is usually used with "!", "%", and "+". For example, "408(12)+" can match 40812 or 408121212. But it cannot match 408. That is, "12" can appear continuously and it must appear at least once.
outband Use outband to configure out-of-band DTMF transmission. Use undo outband to restore the default. Syntax outband { h225 | h245 | nte } undo outband Default The inband DTMF transmission mode is adopted. Views POTS/VoIP voice entity view Default command level 2: System level Parameters h225: Adopts DTMF H.225 out-of-band transmission. h245: Adopts DTMF H.245 out-of-band transmission. nte: Adopts DTMF named telephone event (NTE) transmission.
Default The voice packetization period for g971 is 20 milliseconds, and that for g723, g726, and g726 is 30 milliseconds. Views POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view Default command level 2: System level Parameters g711: Packetization period in milliseconds for g711alaw or g711ulaw codec, an integral multiple of 10 in the range of 10 to 30, with a default of 20.
register-number Use register-number to enable the VoIP gateway to register numbers of a voice entity with an H.323 gatekeeper or SIP server. Use undo register-number to disable a gateway from registering numbers of a voice entity with an H.323 gatekeeper or SIP server. Syntax register-number undo register-number Default After configured with SIP-registration related parameters, a POTS voice entity initiates registration to the SIP server.
Default command level 2: System level Examples # Clear calling statistics on the CMC module. reset voice cmc statistic Related commands display voice cmc reset voice ipp statistic Use reset voice ipp statistic to reset IPP statistics. Syntax reset voice ipp statistic Views User view Default command level 2: System level Examples # Clear IPP statistics.
rtp payload-type nte Use rtp payload-type nte to configure the payload type field in RTP packets in the case of DTMF relay using NTE. Use undo rtp payload-type nte to restore the default. Syntax rtp payload-type nte value undo rtp payload-type nte Default The payload type field in RTP packets is set to 101 in the case of DTMF relay using NTE.
Default command level 2: System level Parameters value: Specifies the timeout interval in seconds for RTP streams, in the range of 10 to 300. Examples # Set the timeout interval for RTP streams to 60 seconds. system-view [Sysname] voice-setup [Sysname-voice] rtp-detect timeout 60 send-ring Use send-ring to enable the local end to play ringback tone. Use undo send-ring to disable the local end from playing ringback tone.
Syntax shutdown undo shutdown Default The voice entity management status is UP. Views POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view Default command level 2: System level Usage guidelines Running command shutdown will cause the voice entity unable to make calls. Examples # Change the management status of voice entity 4 to DOWN.
During actual configuration, it is only necessary to fulfill this command for the VoIP voice entity at the calling gateway. Examples # Enable the tunnel function for VoIP voice entity 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] fast-connect [Sysname-voice-dial-entity10] tunnel-on Related commands • fast-connect • outband • voip called-tunnel enable • voip called-start vad-on Use vad-on to enable VAD.
The VAD discriminates between silence and speech on a voice connection according to signal energies. VAD reduces the bandwidth requirements of a voice connection by not generating traffic during periods of silence in an active voice connection. Speech signals are generated and transmitted only when an active voice segment is detected. Researches show that VAD can save the transmission bandwidth by 50%. Examples # Enable VAD on POTS voice entity 10.
[Sysname] quit reboot # After the router is rebooted, execute the display device verbose command to view the settings. display device verbose Slot No. Board Type Status Max Ports 0 MSR50-60 RPU Board Normal 5 2 SIC-1VT1 Normal 1 Slot 0 Status: Normal Type: MSR50-60 RPU Board Hardware: 3.0 Driver: 1.0 CPLD: 2.0 SD701: 2.0 VCPM: Normal [PCB VER: 3.0 CPLD VER: 1.0 FPGA VER: 4.0] VPM1: Normal [PCB VER: 3.0 CPLD VER: 2.0 DSP VER: 4.0] CPLD VER: 2.
Syntax voice-setup undo voice-setup Views System view Default command level 2: System level Examples # Enter voice view and enable voice services. system-view [Sysname] voice-setup voip called-tunnel enable Use voip called-tunnel enable to enable the tunnel function on the called gateway. Use undo voip called-tunnel enable to disable the tunnel function on the called gateway.
Default The fast connection mode is used for call initialization. Views Voice view Default command level 2: System level Parameters fast: The called GW initializes calls in a fast way. normal: The called GW initializes calls in a non-fast way. Usage guidelines As the process of faculty negotiation is omitted in fast connection procedures, the faculties of the two parties are determined by the GW.
Parameters voip-to-pots time: Specifies the time duration in seconds for switching from the current VoIP link to another VoIP link or a PSTN link (the call backup switching time) in case of a VoIP call failure, in the range of 3 to 30. Usage guidelines For more information about call backup, see Voice Configuration Guide. Examples # Set the time duration for switching from the current VoIP link to another VoIP link or a PSTN link in case of a VoIP call failure to 3 seconds.
Keyword DSCP value in binary DSCP value in decimal af23 010110 22 af31 011010 26 af32 011100 28 af33 011110 30 af41 100010 34 af42 100100 36 af43 100110 38 cs1 001000 8 cs2 010000 16 cs3 011000 24 cs4 100000 32 cs5 101000 40 cs6 110000 48 cs7 111000 56 default 101110 46 ef 101110 46 Usage guidelines The function of this command is the same as the command used for setting DSCP in the "QoS" part of this manual.
Default command level 2: System level Parameters buffer-time time: Specifies the duration in milliseconds of monitoring DSP buffered data. The value for time can be 0 or any number in the range of 180 to 480. Usage guidelines Duration greater than 240 milliseconds is recommended because too small a duration value will result in poor voice quality in the case of severe jitter. Examples # Set the duration of monitoring DSP buffered data to 300 milliseconds.
Voice subscriber line configuration commands The voice subscriber line in this chapter refers to a digital or analog subscriber line, unless otherwise specified. Analog voice subscriber line configuration commands area NOTE: This command applies to 2-wire loop trunk subscriber line FXO only. Use area to configure the type of busy tone for FXO voice subscriber line. Use undo area to restore the default type.
system-view [Sysname] voice-setup [Sysname-voice] area north-america audio-input-gain Use audio-input-gain to set the input gain on a voice interface. Use undo audio-input-gain to restore the default. Syntax audio-input-gain value undo audio-input-gain Default The input gain on a voice interface is 27.5 dB. Views Music on hold (MoH) voice subscriber line view Default command level 2: System level Parameters value: Value of the audio input gain, in the range of 19.5 to 41.
Parameters bandwidth-value: Expected bandwidth for the voice subscriber line, in the range of 1 to 4294967295 kbps. Usage guidelines Obtain the expected bandwidth of a voice subscriber line by querying the ifspeed value of the MIB node with third-party software. The expected bandwidth set by the bandwidth command is used by the network management systems for monitoring bandwidth, but does not affect the actual bandwidth of the interface.
system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] busytone-t-th 3 calling-name Use calling-name to configure the calling name. Use undo calling-name to remove the calling name. Syntax calling-name text undo calling-name Default No calling name is configured.
Default command level 2: System level Examples # Enable CID on voice subscriber line 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid display cid receive Use cid receive to enable CID. Use undo cid receive to disable CID. Syntax cid receive undo cid receive Default CID is enabled.
Syntax cid ring { 0 | 1 | 2 } [ times ] undo cid ring Default CID check is performed between the first and the second rings, and the FXO line goes off-hook as soon as the check completes, cid ring 1 0. Views Analog FXO voice subscriber line view Default command level 2: System level Parameters 0: CID check is performed before the phone rings. 1: CID check is performed between the first and the second rings. 2: CID check is performed between the second and the third rings.
it or it is configured in the number template for the voice entity associated with the FXO voice subscriber line. Examples # Disable voice subscriber line 1/0 from sending calling identity information to the IP network. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] undo cid send cid standard type Use cid standard-type to configure the standard for the FXS voice subscriber line to send messages carrying the calling number information.
Default MDMF is adopted. Views Analog FXS voice subscriber line view Default command level 2: System level Parameters complex: Calling identity information is transmitted in multiple data message format (MDMF). simple: Calling identity information is transmitted in single data message format (SDMF). Usage guidelines Two formats are available: MDMF and SDMF. If the remote end supports one format only, you must use the same message format at the local end.
[Sysname-subscriber-line1/0] undo cng-on Related commands • line • vad-on cptone country-type Use cptone country-type to configure the current device to play the call progress tones of a specified country or region or play the customized call progress tones. Use undo cptone country-type to restore the default. Syntax cptone country-type locale undo cptone country-type Default The call progress tones of China are specified.
Code Country name (including customization) 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 P
Code Country name (including customization) 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 Usage guidelines The cptone country-type CS command enables call progress tones that have been customized with the vi-card cptone-custom command.
Default The amplitude is 1000 for busy tone and congestion tone, 400 for dial tone and special dial tone, and 600 for ringback tone and waiting tone. Views Voice view Default command level 2: System level Parameters all: All types of call progress tones. busy-tone: Busy tone. congestion-tone: Congestion tone. dial-tone: Dial tone. ringback-tone: Ringback tone. special-dial-tone: Special dial tone. waiting-tone: Waiting tone. amplitude value: Amplitude of a progress tone, in the range of 200 to 1500.
Examples # Restore the default settings for voice subscriber line 8/0. system-view [Sysname] subscriber-line 8/0 [Sysname-subscriber-line8/0] default This command will restore the default settings. Continue? [Y/N]:y default subscriber-line Use default subscriber-line to configure the default receiving or transmitting gain on subscriber lines. Use undo default subscriber-line to restore the default value for all voice subscriber lines.
Default The delay signal duration is 400 milliseconds. Views E&M voice subscriber line view Default command level 2: System level Parameters hold milliseconds: Specifies delay signal duration (in milliseconds) in the delay start mode, in the range of 100 to 5000. Examples # Set the delay signal duration in the delay start mode to 500 seconds.
[Sysname-subscriber-line5/0] em-signal delay [Sysname-subscriber-line5/0] delay rising 700 Related commands em-signal delay send-dtmf Use delay send-dtmf to configure a delay before the originating side sends DTMF signals in the immediate start mode. Use undo delay send-dtmf to restore the default. Syntax delay send-dtmf milliseconds undo delay send-dtmf Default The delay before the originating side sends DTMF signals in the immediate start mode is 300 milliseconds.
Views E&M voice subscriber line view Default command level 2: System level Parameters send-wink milliseconds: Specifies an interval (in milliseconds) from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode. The value range for milliseconds is 100 to 5000. Examples # Set the interval from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode to 700 milliseconds.
Related commands em-signal delay wink-rising Use delay wink-rising to configure a maximum amount of time the originating side waits for a wink signal after sending a seizure signal in the wink start mode. Use undo delay wink-rising to restore the default. Syntax delay wink-rising milliseconds undo delay wink-rising Default The maximum amount of time the originating side waits for a wink signal after sending a seizure signal is 3,000 milliseconds in the wink start mode.
Views FXS voice subscriber line view, FXO voice subscriber line view Default command level 2: System level Parameters seconds: Dial delay in seconds, in the range of 0 to 10. Examples # Set the dial delay on FXS subscriber line 1/0 to 5 seconds. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] delay start-dial 5 description (voice subscriber line view) Use description to configure a subscriber line description string. Use undo description to delete the description.
Syntax disconnect lcfo undo disconnect lcfo Default The sending of pulse signal at hangup is disabled, and the system plays busy tones to the other end. Views FXS voice subscriber line view Default Level 2: System level Parameters None Examples # Enable the sending of pulse signals at hangup on the FXS voice subscriber line 5/1.
Examples Actual output might vary depending on the device model. # Display the configuration information about E&M voice subscriber line 5/0. display voice subscriber-line 5/0 Current information ----- subscriber-line5/0 Type = Analog E&M Immediate-Start Status = UP Call Status = BUSYTONE Description = subscriber-line5/0 Interface Private Line = None Cng = Enable Echo Canceller = Enable Echo Canceller Tail-Length = 32 Nlp On = Enable Receive Gain = 0.0 Transmit Gain = 0.
LastSndPacketLen = 0 CmdInBuff = 0 CmdInTotalBuff = 0 DataInBuff = 0 DataInTotalBuff = 0 AbortCmdCount = 0 AbortPktsCount = 0 G723R53ToR63Packet = 0 G723R63ToR53Packet = 0 ClearDspBuffCount = 0 Table 24 Command output Field Description Type Type of voice subscriber line. Status Status of voice subscriber line. Call Status Call status of voice subscriber line. Description Description of voice subscriber line. Private-line Private line dial number of voice subscriber line.
Field Description OutBytes Bytes of sent packets on the voice interface. LastRcvPacketLen Length of the last received packet on the voice interface. LastSndPacketLen Length of the last sent packet on the voice interface. CmdInBuff Number of commands in the command buffer of the voice interface. CmdInTotalBuff Total number of commands in the command buffers of the voice interface card. AbortCmdCount Number of command packets discarded on the voice interface.
Field Description Status of the voice subscriber line: • ADM—The subscriber line is shut down by using the shutdown command. • DOWN—The subscriber line is administratively up but physically down (possibly because no physical link is present or the link has failed). Phy-status • UP(S)—The subscriber line is up at the link layer, but its link is an on-demand link or not present at all. This attribute is typical of BSV interfaces. • UP—The subscriber line is up.
Views Voice view Default command level 2: System level Parameters value: DTMF amplitude in 0.1 dBm increments, in the range of –9.0 to –7.0. Usage guidelines The configuration will apply to the whole device once you carry out this command. Examples # Configure the DTMF amplitude to –8.0 dBm. system-view [Sysname] voice-setup [Sysname-voice] dtmf amplitude -8.
frequency-tolerance value: Absolute frequency deviation (in percentage) when the DTMF detection sensitivity level is set to medium. The value is in the range 1.0 to 5.0 and defaults to 2.0. The greater the value, the higher the probability of false detection. Examples # Set the DTMF detection sensitivity level of voice subscriber line 1/0 to high.
Syntax dtmf threshold analog index value undo dtmf threshold analog index Views Analog FXS voice subscriber line view, analog FXO voice subscriber line view, analog E&M voice subscriber line view Default command level 2: System level Parameters analog: Analog voice subscriber line. index: Index number corresponding to a threshold, an integer 0 through 12. value: Threshold corresponding to the specified index. The value range varies with indexes. For details, see Table 26.
Index Meaning Value range Remarks 5 Upper limit of the ratio of the second largest energy level from the column frequency group to COLMAX. The ratio must be lower than this limit for the input signal to be recognized as a DTMF digit. –18 to –3 dB, with a default of –9 dB The smaller the value, the higher the detection specificity and the lower the detection sensitivity. 6 Upper limit of ROW2nd/ROWMAX. An input signal is recognized as a DTMF digit only when ROW2nd/ROWMAX < 6.
[Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] dtmf threshold analog 9 40 echo-canceller Use echo-canceller to enable echo cancellation and set the echo duration. Use undo echo-canceller to disable the EC function. Syntax echo-canceller { enable | tail-length milliseconds } undo echo-canceller { enable | tail-length } Default The EC function is enabled.
Syntax echo-canceller parameter { convergence-rate value | max-amplitude value | mix-proportion-ratio value | talk-threshold value } undo echo-canceller parameter { convergence-rate | max-amplitude | mix-proportion-ratio | talk-threshold } Default The convergence rate of comfort noise amplitude is 0, the maximum amplitude of comfort noise is 256, the comfort noise mixture proportion control factor is 100, and the threshold of two-way talk is 1.
Parameters busytone: Specifies busy tones as the analog E&M line failure tone and the device plays the busy tones five times. silence: Specifies silence mode. Examples # Configure busy tones as the analog E&M line failure tone failure tone. system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-failure busytone Related commands • open-trunk • em-passthrough em-phy-parm Use em-phy-parm to configure a wire scheme for the analog E&M subscriber line.
Use undo em-signal to restore the default start mode. Syntax em-signal { delay | immediate | wink } undo em-signal Default The immediate start mode is selected for the analog E&M subscriber line.
system-view [Sysname] subscriber-line 6/0 [Sysname-subscriber-line6/0] em-passthrough hookoff-mode Use hookoff-mode to configure the off-hook mode for the FXO voice subscriber line. Use undo hookoff-mode to restore the default. Syntax hookoff-mode { delay | immediate } undo hookoff-mode Default The FXO voice subscriber line operates in the immediate off-hook mode.
Parameters fxs_subscriber_line: FXS voice subscriber line bound to the FXO voice subscriber line. ring-immediately: Specifies the immediate ringing mode. Usage guidelines After an FXS voice subscriber line is bound to the FXO voice subscriber line, the off-hook/on-hook state of these two lines will be consistent.
system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] hookoff-time 500 impedance Use impedance to set the current electrical impedance on an FXO or FXS voice subscriber line. Use undo impedance to restore the default. Syntax impedance { country-name | R550 | R600 | R650 | R700 | R750 | R800 | R850 | R900 | R950 } undo impedance Default The electrical impedance on the FXO or FXS voice subscriber line is the impedance value corresponding to China.
[Sysname-subscriber-line1/0] impedance r600 nlp-on Use nlp-on to enable the EC nonlinear processing function on a voice interface. Use undo nlp-on to disable the function. Syntax nlp-on undo nlp-on Default The EC nonlinear processing function is enabled. Views Voice subscriber line view Default command level 2: System level Usage guidelines This command takes effect only after the echo-canceller enable command is configured.
mirror (analog voice subscriber line view) Use mirror to mirror the PCM, RTP, or voice command data on the analog voice subscriber line to a specified interface or destination. Use undo mirror to remove a mirror entry or all mirror entries. Syntax mirror number number { pcm | { in | out | all } { command | data } } to { local-interface interface-type interface-number [ mac H-H-H ] | remote-ip ip-address [ port port ] } undo mirror [number number] Default No traffic is mirrored.
open-trunk Use open-trunk to enable E&M non-signaling mode. Use undo open-trunk to disable the E&M non-signaling mode. Syntax open-trunk { caller monitor interval | called } undo open-trunk Default The E&M non-signaling mode is disabled.
Parameters general: Uses the universal frame erasure algorithm. specific: Uses the specific algorithm provided by the voice gateway. Examples # Configure the voice gateway to use the universal packet loss compensation algorithm. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] plc-mode general receive gain Use receive gain to set the gain value at the voice subscriber line input end. Use undo receive gain to restore the default.
reset voice cmc statistic Use reset voice cmc statistic to clear calling statistics on the CMC module. Syntax reset voice cmc statistic Views User view Default command level 2: System level Examples # Clear calling statistics on the CMC module. reset voice cmc statistic Related commands display voice cmc reset voice ipp statistic Use reset voice ipp statistic to reset IPP statistics.
Examples # Clear IVA statistics. reset voice iva statistic Related commands display voice iva statistic ring-detect debounce Use ring-detect debounce to configure the debounce time of ring detection on a FXO subscriber line. By setting different debounce times, you can detect ring signals of different frequencies and waveforms. 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.
ring-detect frequency Use ring-detect frequency to set the frequency value for ring detection. Use undo ring-detect frequency to restore the default. Syntax ring-detect frequency value undo ring-detect frequency Default The frequency for ring detection is 40 Hz. Views Analog FXO voice subscriber line view Default command level 2: System level Parameters value: Frequency value for ring detection, in Hz. The value is in the range 30 to 100 with the step of 10.
Parameters enable: Enables busy-tone sending on the FXO interface. time seconds: Duration of busy tone in seconds, in the range of 2 to 15. It defaults to 3 seconds. This parameter is available after you enable busy tone sending with the send-busytone enable command. Examples # Enable FXO interface 1/0 to send busy tone that lasts 5 seconds.
undo silence-th-span Default The silence threshold is 20 and the silence duration for automatic on-hook is 7,200 seconds (2 hours). Views Analog FXO subscriber line view Default command level 2: System level Parameters threshold: Silence threshold value in the range of 0 to 200. If the amplitude of voice signals from the switch is smaller than this value, the system regards the voice signals as silence. Generally, the signal amplitude on the links without traffic is in the range of 2 to 5.
[Sysname-subscriber-line5/0] slic-gain 1 subscriber-line Use subscriber-line to enter the specified voice subscriber line view. Syntax subscriber-line line-number Views System view Default command level 2: System level Parameters line-number: Voice subscriber line number. Examples # Enter the view of the voice subscriber line 1/0 in system view.
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 time duration for the sending of the pulse signals at hangup. Use undo timer disconnect-pulse to restore the default setting.
Usage guidelines Upon the expiration of the timer, the subscriber will be prompted to hook up and the call will be terminated. Examples # Set the maximum interval between off-hook and dialing the first digit to 10 seconds. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer first-dial 15 timer hookflash-detect Use timer hookflash-detect to configure the time range for the duration of an on-hook condition that will be detected as a hookflash.
Views Analog FXO voice subscriber line view Default command level 2: System level Parameters milliseconds: Interval between on-hook and off-hook in milliseconds, in the range of 500 to 4000. Usage guidelines In the delay off-hook mode, the on-hook/off-hook state of FXS and FXO voice subscriber lines is consistent. When an FXS voice subscriber line goes off-hook, the FXO voice subscriber line to which the FXS voice subscriber line is bound goes off-hook, too.
timer wait-digit Use timer wait-digit to configure the maximum time duration the system waits for a digit. Use undo timer wait-digit to restore the default time settings. Syntax timer wait-digit { seconds | infinity } undo timer wait-digit Default The maximum time duration the system waits for a digit is 5 seconds. Views E&M voice subscriber line view Default command level 2: System level Parameters seconds: Maximum duration in seconds the system waits for a digit, in the range of 3 to 600.
Usage guidelines This command is applicable to FXO, FXS, E&M, BSV and E1/T1 voice subscriber lines. When the voice signal power is small on the output line, use this command to increase the transmit gain. Gain adjustment might lead to call failures. Adjusting the gain is not recommended. If necessary, make sure you understand the impact of the adjustment on your call before you use the command. Examples # Set the voice output gain value to –6.7dB on subscriber line 1/0.
vi-card busy-tone-detect NOTE: This command applies to the FXO interface only. Use vi-card busy-tone-detect to configure the parameters for the busy tone detection on the FXO interface. Use undo vi-card busy-tone-detect to restore the default settings.
Examples # Enable the automatic busy tone detection on subscriber line 2/0, with the busy tone index being 0. system-view [Sysname] voice-setup [Sysname-voice] vi-card busy-tone-detect auto 0 2/0 # Manually configure busy tone parameter indexed as 1, including duration limit of high/low level, tolerance of high/low level duration, and duration difference of high/low level.
time1: Make time for the first make-to-break ratio in milliseconds, in the range of 30 to 8192. In the case of continuous play, the value is 8192. time2: Break time for the first make-to-break ratio in milliseconds, 30 through 8191. time3: Make time for the second make-to-break ratio in milliseconds, 30 through 8191. time4: Break time for the second make-to-break ratio in milliseconds, 30 to 8191.
• display device (Fundamentals Command Reference) Digital voice subscriber line configuration commands amd enable Use amd enable to enable the answering machine detection (AMD) function. Use undo amd enable to disable the AMD function. Syntax amd enable undo amd enable Default The AMD function is disabled. Views Digital voice subscriber line view Default command level 2: System level Examples # Enable the AMD function on voice subscriber line 1/0:1.
Views Voice view Default command level 2: System level Parameters machine-time value: Sets the answering machine recognition time in milliseconds. The value must be multiples of 10 in the range of 10 to 60000. If the greeting of the called party lasts longer than the answering machine recognition time, the called party will be considered as an answering machine. max-analyze-time value: Sets the maximum time in milliseconds for the AMD function to analyze the voice of the speaker.
Default command level 2: System level Parameters all: Specifies the remote end to send the category of the calling party and calling number. ka: Specifies the remote end to send only the category of the calling services. Usage guidelines Configure the local end with this command to support automatic number identification. In normal situations, configure the all keyword. Use the ka keyword only when required by the connected switch to prevent call failures.
Usage guidelines When the number of collected digits is smaller than the set value, the system will wait for the next digit until the timer expires. When the number of collected digits equals to or exceeds the set value, the system will request the calling party information. Before you can configure this command, you must configure the ani command. Examples # Set the number of collected digits before the requesting of the calling party information to three.
Examples # Configure the originating side to disable the terminating side from sending answer 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-cas1/0:0] undo answer enable Related commands • re-answer enable • timer dl re-answer callmode Use callmode to configure the connection mode for an R2 call. Use undo callmode to restore the default setting.
Views E1 interface view, T1 interface view Default command level 2: System level Parameters ts-set-number: Number of a created timeslot (TS) set, in the range of 0 to 30 (for E1 interfaces) or 0 to 23 (for T1 interfaces). Usage guidelines You can configure signaling parameters in signaling view. The TS set number specified in this command must already be created with the timeslot-set command. After entering a signaling view, you can configure signaling parameters as desired.
Views R2 CAS view Default command level 2: System level Usage guidelines In some countries, if the terminating side controls circuit reset in the R2 signaling exchange process, when the calling party disconnects a call and the originating side sends a clear-forward signal to the terminating side, the terminating side sends a clear-back signal as an acknowledgement, and then sends a release guard signal to indicate that the line of the terminating side is released.
begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters. Usage guidelines If no subscriber lines are specified, the command displays information about all subscriber lines (analog and digital subscriber lines).
Field Description Type Signaling type on the voice subscriber line. Status Status of the voice subscriber line. Call Status Status of the voice protocol call. Description Information about the voice subscriber line. Private Line Private line dialup mode of the voice subscriber line. Cng Comfort noise setting on the voice subscriber line. The subscriber line's description Description of the subscriber line. Echo Canceller Echo cancellation setting on the voice subscriber line.
Field Description Type of the voice subscriber line or signaling used by the voice subscriber line: Type • • • • • • • • • • • PAGE—PAGE subscriber line. FXS-LS—FXS subscriber line that uses loop-start signaling. FXO-LS—FXO subscriber line that uses loop-start signaling. FXS-GS—FXS subscriber line that uses ground-start signaling. FXO-GS—FXO subscriber line that uses ground-start signaling. R2—Subscriber line that uses R2 signaling. ISDN—Subscriber line that uses ISDN signaling.
dl-bits NOTE: This command applies to R2 signaling only. Use dl-bits to configure the ABCD bit pattern for R2 signals. Use undo dl-bits to restore the defaults.
Signal Default rx-bits ABCD Default tx-bits ABCD Release-guard 1001 1001 Usage guidelines You can use this command to accommodate to the ABCD bit pattern schemes used in different countries. Examples # Set the ABCD bit pattern to 1101 for received R2 idle signal, and to 1011 for transmitted R2 idle signal.
Related commands timer dtmf dtmf threshold digital Use dtmf threshold digital to set the DTMF detection sensitivity. Use undo dtmf threshold digital to restore the default DTMF detection sensitivity. Syntax dtmf threshold digital value undo dtmf threshold digital Default The DTMF detection sensitivity level is 0, that is, insensitive. Views BSV voice subscriber line view Default command level 2: System level Parameters digital: Sets a digital voice subscriber line. value: 0 or 1.
Views BSV BRI interface view Default command level 2: System level Examples # Disable interface BSV BRI 2/0 from generating linkUp/linkDown traps. system-view [Sysname] interface bri 2/0 [Sysname-Bri2/0] undo enable snmp trap updown final-callednum enable NOTE: This command applies to R2 signaling only. Use final-callednum enable to enable the originating side to send a number terminator to the terminating side after it sends all digits of a called number.
force-metering enable NOTE: This command applies to R2 signaling only. Use force-metering enable to enable R2 metering signal processing. Use undo force-metering enable to disable R2 metering signal processing. Syntax force-metering enable undo force-metering enable Default R2 metering signal processing is disabled.
Views R2 CAS view Default command level 2: System level Usage guidelines You can use the undo form of this command to accommodate to the R2 interregister signaling in some countries where Group B signals is not supported or cannot be interpreted correctly. Examples # Adopt Group B signals to complete registers exchange.
Examples # Associate a POTS entity with a TS set on an E1 interface. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] line 1/0:1 Related commands • entity • pri-set • timeslot-set link-delay Use link-delay to set the physical state change suppression interval on an E1/T1 interface. Use undo link-delay to restore the default.
Examples # Set the physical state change suppression interval to 2 seconds on the E1/T1 interface E1 2/0, so that the system can detect a physical state change and report the change 2 to 7 seconds after the change occurs.
calling calling-number: Specifies a calling number, a string of up to 31 characters that can only contain 0 through 9. Traffic matching the specified calling number is mirrored. bdsp: Mirrors the PCM data of the back-end DSP of the specified time slot. fdsp: Mirrors the PCM data of front-end DSP of the specified time slot. command: Mirrors voice command data. channel-number: Specifies a time slot in the range of 0 to 29. The data on the specified time slot will be mirrored.
Default command level 2: System level Parameters zone-name: Country or region name. The argument can be one of the following values: • argentina: Uses Argentinean R2 signaling standard. • australia: Uses Australian R2 signaling standard. • bengal: Uses Bengalese R2 signaling standard. • brazil: Uses Brazilian R2 signaling standard. • china: Uses Chinese R2 signaling standard. • custom: Uses customized R2 signaling standard. • hongkong: Uses Hongkong R2 signaling standard.
Related commands • register-value • force-metering enable pcm Use pcm to configure a companding law used for quantizing signals. Use undo pcm to restore the default. Syntax pcm { a-law | μ-law } undo pcm Default The companding law for VE1 interfaces is A-law, while that for VT1 interfaces is μ-law.
Default No PRI group is created. Views E1 interface view, T1 interface view Default command level 2: System level Parameters range: Specifies timeslots to be bundled. Timeslots are numbered 1 through 31 on an E1 interface and 1 to 24 on a T1 interface. You can specify a single timeslot by specifying a number, a range of timeslots by specifying a range in the form of number1-number2, or several discrete timeslots by specifying number1, number2-number3.
Default The QSIG tunneling function is disabled. Views Digital voice subscriber line view Default command level 2: System level Examples # Enable the QSIG tunneling function. system-view [Sysname] subscriber-line 1/0:15 [Sysname-subscriber-line1/1:15] qsig-tunnel enable re-answer enable NOTE: This command applies to R2 signaling only. Use re-answer enable to enable the originating side to support re-answer signal processing. Use undo re-answer enable to restore the default.
Related commands • answer enable • timer dl re-answer register-value NOTE: This command applies to R2 signaling only. Use register-value to configure R2 register signal values. Use undo register-value to restore the defaults.
req-callednum-and-switchgroupa value: Specifies the send last digit and changeover to Group A signal value in the range of 1 to 16. req-callingcategory value: Specifies the send calling category signal value in the range of 1 to 16. req-currentcallednum-in-groupc value: Specifies the send current called number signal in Group C state, in the range of 1 to 16. req-currentdigit value: Specifies the send current digit signal value in the range of 1 to 16.
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-cas1/0:0] register-value req-callingcategory 7 Related commands group-b enable renew NOTE: This command applies to R2 signaling only. Use renew to configure the values of C bit and D bit in R2 signaling. Use undo renew to restore the default. The default value varies with R2 signaling standards in countries.
reverse NOTE: This command applies to R2 signaling only. Use reverse to configure line signal inversion mode. Use undo reverse to invert ABCD bits of the current line signaling whose values are "1" after the reverse command is executed. Syntax reverse ABCD undo reverse Views R2 CAS view Default command level 2: System level Parameters ABCD: Indicates whether corresponding ABCD bits in R2 signaling need inversion.
Syntax seizure-ack enable undo seizure-ack enable Default The originating side requires the terminating side to send seizure acknowledgement signal. Views R2 CAS view Default command level 2: System level Usage guidelines Generally, the terminating side acknowledges received seizure signals. The R2 line signaling coding schemes in some countries however do not require the terminating side to do this.
maxpoll: Selects the timeslot with the greatest number from available timeslots in the first timeslot polling. In later pollings, it selects in descending order timeslots with numbers less than the one picked out in the previous polling. Suppose TS31 and TS29 are not available. In the first polling, TS30 will be picked out for use and in the next polling, TS28. min: Selects the timeslot with the lowest number from available timeslots.
[Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 5 [Sysname-cas1/0:5] sendring ringbusy enable Related commands timer ring signal-value Use signal-value to configure the ABCD bit patterns of idle receive, receive seized, idle transmit, and transmit seized signals on the digital E&M voice subscriber line. Use undo signal-value to restore the defaults.
special-character NOTE: This command applies to R2 signaling only. Use special-character to configure the special characters acceptable during register signal exchange. Use undo special-character to remove the configured special characters. Syntax special-character character number undo special-character character number Default No special characters are configured.
Default command level 2: System level Parameters slot-number: Number of the voice subscriber line automatically created upon creation of a TS set or ISDN PRI group. ts-set-number: Number of the TS set that has been created. 15: Indicates the subscriber line is created for the ISDN PRI group created on an E1 interface. 23: Indicates the subscriber line is created for the ISDN PRI group created on a T1 interface.
Parameters internal: Specifies the internal crystal oscillator clock, which can be the one on the main board or the one on the E1/T1 card, as the clock source of the E1/T1 interface. Note that the SIC card does not have a crystal oscillator clock. In this case, the E1/T1 interface is the master in time synchronization. line: Specifies the clock on the peer device as the clock source of the E1/T1 interface. In this case, the E1/T1 interface is a client in time synchronization.
Default command level 2: System level Parameters answer time: Timeout time in milliseconds of R2 answer signal, in the range of 100 to 120000 with a default of 60000. After the originating side sends a seizure acknowledgement signal, the terminating side should send back an answer signal within the timeout time. If the terminating side fails to send an answer signal within the timeout time, the originating side will clear the connection.
Syntax timer dtmf time undo timer dtmf Default The delay is 50 milliseconds. Views R2 CAS view Default command level 2: System level Parameters time: Delay before sending a DTMF signal in milliseconds, in the range of 50 to 10000. Usage guidelines Before you can configure this command, you must configure the dtmf enable command. Generally, the originating side starts sending DTMF signals immediately after receiving a line seizure acknowledgement signal.
Default command level 2: System level Parameters seconds: Interval (in seconds) at which the interface sends keepalive packets, in the range 0 to 32767. Examples # Set the keepalive interval to 100 seconds for interface BSV BRI 2/0. system-view [Sysname] interface bri 2/0 [Sysname-Bri2/0] timer hold 100 timer register-pulse persistence NOTE: This command applies to R2 signaling only.
Related commands timer register-complete group-b timer register-complete group-b NOTE: This command applies to R2 signaling only. Use timer register-complete group-b to configure the timeout value of R2 group B signals. After the terminating side switch to Group B, it should send Group B signals within this time period. Use undo timer register-complete group-b to restore the default timeout value of R2 group B signals.
undo timer ring { ringback | ringbusy } Default The duration of playing the ringback tone is 60,000 milliseconds and that of playing the busy tone is 30,000 milliseconds. Views R2 CAS view Default command level 2: System level Parameters ringback time: Sets the duration in milliseconds of playing ringback tone, in the range of 1000 to 90000. ringbusy time: Sets the duration in milliseconds of playing busy tone, in the range of 1000 to 90000.
timeslots-list: Timeslot range. Timeslots are numbered 1 through 31 for an E1 interface and 1 through 24 for a T1 interface. TS 16 for an E1 interface (or TS24 for a T1 interface) is used to transmit control signaling. signal: Specifies a signaling mode for the TS set, which should be consistent with that adopted by the central office. It includes the following types of signaling: • e&m-delay: Adopts the delay start mode of digital E&M signaling.
Parameters timeslots-list: Timeslot range. Timeslots are numbered 1 through 31 on an E1 interface and 1 through 24 on a T1 interface. You can specify a single timeslot by specifying a number, a range of timeslots by specifying a range in the form of number1-number2, or several discrete timeslots by specifying number1, number2-number3. Examples are 1-14, 15, 17-31. dual: Bidirectional trunk. in: Incoming trunk. out: Outgoing trunk. Usage guidelines This command applies to R2 signaling only.
query: Queries status of the circuits of specified timeslots to see whether the circuits are busy, open, or blocked in real time. reset: Resets the circuits of specified timeslots when they cannot automatically reset. You might need to do this if the state of an administratively blocked or opened circuits cannot recover for example. timeslots timeslots-list: Specifies a timeslot range. Timeslots are numbered 1 through 31 for an E1 interface and 1 through 24 for a T1 interface.
Dial plan configuration commands caller-group Use caller-group to bind a subscriber group to a voice entity. Use undo caller-group to remove the binding of a subscriber group or all subscriber groups to a voice entity. Syntax caller-group { deny | permit } subscriber-group-list-number undo caller-group { { deny | permit } subscriber-group-list-number | all } Default No subscriber group is bound to a voice entity, that is, any calling number is allowed to originate or accept calls.
Use undo caller-permit to remove the configuration. Syntax caller-permit calling-string undo caller-permit { calling-string | all } Default No calling number is configured, that is, outgoing calls are not restricted. Views POTS entity view, VoIP entity view, VoFR entity view, IVR entity view Default command level 2: System level Parameters all: Specifies all calling numbers. calling-string: Specifies the calling number permitted to originate a call to the local voice entity.
Character Meaning Plus sign (+) Indicates the sub-expression before it appears one or more times. However, if a calling number starts with the plus sign, the sign itself does not have special meanings, and only indicates that the following is an effective number and the number is E.164-compliant. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on, and +110022 is an E.164-compliant number.
Use undo description to remove the subscriber group description string. Syntax description text undo description Default No subscriber group description string is configured. Views Subscriber group view Default command level 2: System level Parameters text: Subscriber group description string, consisting of 1 to 80 case-insensitive characters. Usage guidelines The description configured for a subscriber group by using the description command does not affect the use of the subscriber group.
Parameters string: Prefix code, a character string consisting of 1 to 31 characters that can include 0 through 9, comma (,), pound sign (#), and asterisk (*). Table 31 describes these characters: Table 31 Description of characters in the string argument Character Meaning 0-9 Digits 0 through 9. Comma (,) One comma represents a pause of 500 milliseconds and it can be positioned anywhere in a number. Pound sign (#) or asterisk (*) Indicates a valid digit each.
all: Specifies all subscriber groups. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression.
display voice number-substitute Use display voice number-substitute to display the configuration information of a number substitution rule list. Syntax display voice number-substitute [ list-tag ] [ | { begin | exclude | include } regular-expression ] Views Any view Default command level 2: System level Parameters list-tag: Serial number of a number substitution rule list, in the range of 1 to 2147483647. |: Filters command output by specifying a regular expression.
undo dot-match Default The dot match rule is end-only. Views Voice number-substitute view Default command level 2: System level Parameters end-only: Reserves the digits to which all ending dots (.) in the input number correspond. left-right: Reserves from left to right the digits to which the dots in the input number correspond. right-left: Reserves from right to left the digits to which the dots in the input number correspond.
Parameters rule-number: Serial number of a number substitution rule (namely, the serial number of a number substitution rule configured by using the rule command), in the range of 0 to 31. Usage guidelines Use the first-rule to configure the preferred number substitution rule in the current number substitution rule list. In a voice call, the system first uses the rule defined by the first-rule command for number substitution.
string: A string consisting of any characters of digits 0 through 9, and pound sign (#), asterisk (*), dot (.), exclamation point (!), plus sign (+), percent sign (%), brackets ([ ]), parentheses (()), and hyphen (-).The characters in a string are described in the following table: • Table 33 Meanings of characters in a string Character Meaning 0-9 Digits from 0 through 9. Pound sign (#) and asterisk (*) Each represents a valid digit. Wildcard, which can match any valid digit. For example, 555….
Example # Configure the calling number match template 660268 for subscriber group 2. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] subscriber-group 2 [Sysname-voice-dial-group2] match-template 660268 Related commands • description • subscriber-group max-call (voice dial program view) Use max-call to configure maximum-call-connection sets. Use undo max-call to remove the specified maximum-call-connection set or all maximum-call sets.
max-call (voice entity view) Use max-call to bind the maximum-call-connection set to a voice entity. Use undo max-call to remove the binding. You can bind each voice entity to only one maximum-call-connection set. Syntax max-call set-number undo max-call Default No maximum-call-connection set is bound to the voice entity..
Default command level 2: System level Parameters longest: Matches the longest number. shortest: Matches the shortest number. Usage guidelines If the longest-number match mode is configured and the rule command with the input-format argument ending in a dollar sign ($) is carried out, after a user dials a number, the system does not look up the voice entity to connect the call until the dialing interval expires.
number-substitute Use number-substitute to create a number substitution rule list and enter voice number-substitute view. Use undo number-substitute to remove a specified number substitution rule 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.
Default command level 2: System level Parameters priority-order: Priority of a voice entity, in the range of 0 to 10. The smaller the value, the higher the priority. Usage guidelines If you have configured priority levels for voice entities and the selection priority rules (see the select-rule commands), the router selects the voice entity with the highest priority to initiate a call. Examples # Set the priority level of voice entity 10 to 5.
[Sysname] subscriber-line1/0 [Sysname-subscriber-line1/0] private-line 5559262 rule Use rule to configure a number substitution rule. Use undo rule to remove a specified number substitution rule or all number substitution rules. Syntax rule rule-tag input-number output-number [ number-type input-number-type output-number-type | numbering-plan input-numbering-plan output-numbering-plan ] * undo rule { rule-tag | all } Default No number substitution rule is configured.
Character Meaning Plus sign (+) The character or sub-expression before the plus sign can appear one or more times. However, if a calling number starts with the plus sign, the sign itself does not have special meanings, and only indicates that the following is an effective number and the number is E.164-compliant. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on, and +110022 is an E.164 number.
Table 35 Input number type Number type Description abbreviated Abbreviated number. any Any number. international International number. national National number, but not a local network. network Specific service network number. reserved Reserved number. subscriber Local network number. unknown Number of an unknown type. output-number-type: Type of an output number involved in number substitution. For the values, see Table 36.
Table 38 Output numbering plan 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 After you create a number substitution rule list successfully, you need to use this command to configure specific number substitution rules for it.
Related commands • substitute • number-substitute • first-rule • dot-match select-rule rule-order Use select-rule rule-order to configure match order of rules for the voice entity selection. Use undo select-rule rule-order to restore the default. Syntax select-rule rule-order 1st-rule [ 2nd-rule [ 3rd-rule ] ] undo select-rule rule-order Default The match order of rules for the voice entity selection is exact match->voice entity priority->random selection.
• If there are multiple rules, the system first selects a voice entity according to the first rule. • If the first rule cannot decide which voice entity should be selected, the system applies the second rule. If the second rule still cannot decide a voice entity, the system applies the third rule. • If all the rules cannot decide which voice entity should be selected, the system selects a voice entity with the smallest ID.
Examples # Configure the maximum number of voice entities found to 5. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] select-rule search-stop 5 Related commands • select-rule rule-order • select-rule type-first select-rule type-first Use select-rule type-first to configure a rule for voice entity type selection priority. Use undo select-rule type-first to remove a rule for voice entity type selection priority.
Usage guidelines The command is used to configure the sequence of voice entity type selection priority. If different types of voice entities are qualified for a call connection, the system selects a suitable voice entity according to the voice entity type selection priority rule configured by the select-rule type-first command. The order of entering the parameters determines voice entity type priorities.
send-number Use send-number to configure the number sending mode. Use undo send-number to restore the default number sending mode. Syntax send-number { digit-number | all | truncate } undo send-number Default The truncate mode is used. Views POTS entity view Default command level 2: System level Parameters digit-number: Number of digits (that are extracted from the end of a number) to be sent, in the range of 0 to 31. It is not greater than the total number of digits of the called number.
Syntax subscriber-group list-number undo subscriber-group { list-number | all } Default No subscriber group is created. Views Voice dial program view Default command level 2: System level Parameters list-number: Subscriber group ID in the range of 1 to 2147483647. all: Specifies all subscriber groups. Usage guidelines At most ten subscriber groups can be configured for the system. Examples # Enter voice dial program view and create a subscriber group.
Default command level 2: System level Parameters called: Applies the number substitution rule to a called number. calling: Applies the number substitution rule to a calling number. list-number: Serial number of a number substitution rule list configured by using the number-substitute command), in the range of 1 to 2147483647.
Parameters incoming-call: Binds the calling/called number of incoming calls to the number substitution rule list. outgoing-call: Binds the calling/called number of outgoing calls to the number substitution rule list. called: Applies the number substitution rule to a called number. calling: Applies the number substitution rule to a calling number. all: Specifies all number substitution rule lists.
Views Voice dial program view Default command level 2: System level Parameters character: Dial terminator, which can be any of 0 through 9, pound sign (#), or asterisk (*). Usage guidelines Note that if you set the argument character to a pound sign (#) or an asterisk (*), and if the first character of the configured entity number is the same as the argument character (# or *), the device takes this first character as a common number rather than a dial terminator.
SIP configuration commands address sip Use address sip to configure SIP routing for the VoIP voice entity. Use undo address sip to remove specified SIP routing configuration. Syntax address sip { dns domain-name [ port port-number ] | enum-group group-number | ip ip-address [ port port-number ] | proxy | server-group index } undo address sip { dns | ip | proxy } Default No routing policy is configured for the VoIP voice entity.
[Sysname-voice-dial-entity10] address sip dns cc.news.com Related commands address sip server-group call-fallback Use call-fallback register to enable re-registration in the case of a call failure. Use undo call-fallback register to disable call failure-triggered re-registration. Syntax call-fallback register undo call-fallback register Default Call failure-triggered re-registration is disabled.
Parameters ssl-server-policy server-policy-name: References an SSL server policy. The policy name is a string of 1 to 16 case-insensitive characters. ssl-client-policy client-policy-name: References an SSL client policy. The policy name is a string of 1 to 16 case-insensitive characters. Usage guidelines The SSL policies to be referenced must have been configured.
Examples # Display the statistics about all SIP calls.
Table 41 Command output Field Description TPT Message Statistics about SIP transport layer messages, including UDP, TCP, SCTP, and TLS. The messages of each type include InMsg, (received), OutMsgSucc (transmitted successfully), and OutMsgFail (sending failure). Statistics of SIP transaction messages. These messages include: TXN Message • • • • Inv_Cli (INVITE transaction of client). NonInv_Cli (Non-INVITE transaction of client). Inv_Srv (INVITE transaction of server).
display voice sip connection Use display voice sip connection to display information about SIP connections over a specific transport layer protocol, including both established and attempted connections. Syntax display voice sip connection { tcp | tls } [ | { begin | exclude | include } regular-expression ] Views Any view Default command level 1: Monitor level Parameters tcp: Displays the information of all TCP connections. tls: Displays the information of all TLS connections.
Views Any view Default command level 2: System level Parameters all: Displays all ENUM translation rule groups. mark group-number: Displays the specified ENUM translation rule group with a number from 1 to 15. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow.
Default command level 2: System level Parameters dns record: Displays DNS records for SIP. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters. Examples # Display the temporarily saved identifications and contact addresses for SIP users. display voice sip dynamic-contact-address Number Entity Contact address Expires Type +-----------------------------------------------------------------------+ 1000 40001 100.1.1.1:5060 3501 Register 2000 40002 100.1.1.1:5060 20 Transfer 3000 40003 cc.news.
Parameters |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression.
31 88 503 503 32 102 504 504 33 111 500 500 34 127 500 500 Table 45 Command output Field Description PSTN-Reason PSTN release cause code. SIP status code corresponding to a PSTN release cause code. SIP-Status If the configured SIP status code is different from the default, it is highlighted with an asterisk. Default SIP status code corresponding to a PSTN release cause code. Default # Query the SIP status code to PSTN release cause code mappings.
25 487 127 127 26 488 127 127 27 500 41 41 28 501 79 79 29 502 38 38 30 503 41 41 31 504 102 102 32 505 127 127 33 513 127 127 34 600 17 17 35 603 21 21 36 604 1 1 37 606 58 58 Table 46 Command output Field Description SIP-Status SIP status code. PSTN release cause code corresponding to a SIP status code. PSTN-Reason Default If the configured PSTN release cause code is different from the default, it is highlighted with an asterisk.
Usage guidelines If you configure the destination port in the address sip { dns domain-name [ port port-number ] | enum-group group-number }, proxy dns domain-name [ port port-number ], or mwi-server dns domain-name [ port port-number ] command, the DNS lookup mode can only be Type-A. Examples # Set the DNS lookup mode to SRV.
Table 47 Command output Field Description Number User number. Entity Entity number. Registrar Server Address of the registrar, in the format of IP address + port number or domain name + port number. Expires Aging time for a user number in seconds. State in which a number stays: Status • • • • • • offline. online. login. logout. dnsin—DNS query is being performed before the number is registered. dnsout—DNS query is being performed before the number is deregistered.
enum-group Use enum-group to create an ENUM translation rule group. Use undo enum-group to delete an ENUM translation rule group. Syntax enum-group group-number undo enum-group { group-number | all } Default No ENUM translation rule group exists. Views Voice dial program view Default command level 2: System level Parameters group-number: Number of the ENUM translation group, in the range of 1 to 15. all: Deletes all ENUM translation rule groups.
register: Sets the keepalive mode to register. Examples # Set the keepalive mode to options and set the interval for sending options packets to 30 seconds. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] keepalive options interval 30 Related commands redundancy mode line-check enable Use line-check enable to enable checking the status of voice subscriber lines associated with POTS voice entities.
listen transport Use listen transport to enable the listening port for the transport layer protocol. Use undo listen transport to restore the default. Syntax listen transport { tcp | tls | udp } undo listen transport { tcp | tls | udp } Default Both the UDP and TCP listening ports are enabled, and the TLS listening port is disabled.
• transport media-protocol Use media-protocol to specify the media flow protocols for SIP calls. Use undo media-protocol to restore the default. Syntax media-protocol { rtp | srtp } * undo media-protocol Default SIP calls use RTP as the media flow protocol. Views SIP client view Default command level 2: System level Parameters rtp: Specifies the Real-time Transport Protocol (RTP) as the media flow protocol for SIP calls.
Views POTS entity view, VoIP entity view Default command level 2: System level Usage guidelines For more information about DTMF H.225 out-of-band, DTMF H.245 out-of-band, and DTMF NTE transmission modes, see Voice Configuration Guide. Examples # Configure the out-of-band SIP DTMF transmission for VoIP entity 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] address sip ip 10.1.1.
[Sysname-voice-sip] outbound-proxy ipv4 169.54.5.10 port 1120 # Configure domain name abc.com and port number 1100 of the outbound proxy server for the SIP UA. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] outbound-proxy dns abc.com port 1100 privacy Use privacy to add the P-Preferred-Identity or P-Asserted-Identity header field. Use undo privacy to remove the configuration.
Default No proxy server information is configured for SIP UA. Views SIP client view Default command level 2: System level Parameters dns domain-name: Domain name of the proxy server, which consists of character strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters.
PSTN release cause code PSTN release cause description SIP status code SIP status description 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. 19 No answer from user! 480 Temporarily unavailable. 20 Subscriber absent! 480 Temporarily unavailable 21 Call rejected! 403 Forbidden. 22 Number changed! 410 Gone. 23 Redirection to new destination! 410 Gone.
PSTN release cause code PSTN release cause description SIP status code SIP status description 87 User not member of Closed User Group (CUG)! 403 Forbidden. 88 Incompatible destination! 503 Service unavailable. 102 Recovery on timer expiry! 504 Gateway timeout. 111 Protocol error, unspecified! 500 Server internal error. 127 Interworking, unspecified! 500 Server internal error.
SIP status code SIP status description PSTN release cause code PSTN release cause description 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. 79 Service or option not implemented, unspecified! 407 Proxy authentication required. 21 Call rejected! 408 Request timeout. 102 Recovery on timer expiry! 410 Gone.
SIP status code SIP status description PSTN release cause code PSTN release cause description 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.
Examples # Enable the SIP registrar. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] register-enable on redundancy mode Use redundancy mode to set the backup mode. Use undo redundancy mode to restore the default backup mode. Syntax redundancy mode { homing | parking } undo redundancy mode Default The backup mode is parking. Views SIP client view Default command level 2: System level Parameters homing: Sets the backup mode to homing.
Default UDP is adopted. Default The SIP URL scheme is adopted. Default No registrar information is configured on the SIP UA. If you execute this command without providing the transport layer protocol type, the UDP protocol is used during registration. If you execute this command without providing the URL scheme, the SIP URL scheme is used.
[Sysname-voice] sip [Sysname-voice-sip] registrar ipv4 169.54.5.10 port 1120 expires 120 tcp # Specify the domain name cc.news.com, the port number 1100, and the registration aging time 120 seconds of the main registrar. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] registrar dns cc.news.com port 1100 expires 120 remote-party-id Use remote-party-id to add the Remote-Party-ID header field. Use remote-party-id to remove the configuration.
conn-id: Connection ID, in the range 0 to 1499. You can view connection IDs with the display voice sip connection command. Examples # Clear the SIP connection 1 over TCP. reset voice sip connection tcp id 1 reset voice sip dns-record Use reset voice sip dns-record to clear SIP DNS records. Syntax reset voice sip dns-record Views User view Default command level 2: System level Examples # Clear SIP DNS lookup records.
Views ENUM translation rule group view Default command level 2: System level Parameters tag: Sets the number of the ENUM translation rule in the range 1 to 2147483647. You can configure up to eight ENUM translation rules for the group. preference value: Sets the preference value of the ENUM translation rule in the range 1 to 2147483647. The smaller the value, the higher the priority. match-pattern: Telephone number pattern, supporting regular expressions.
[Sysname-voice] sip [Sysname-voice-sip] sip-comp Use sip-comp to configure SIP compatibility. Use undo sip-comp to restore the default. Syntax sip-comp { callee | dt | from | substitute | t38 | x-parameter } * undo sip-comp { callee | dt | from | substitute | t38 | x-parameter } * Default • The destination number is obtained from the request-line, which is the start line in an SIP request message.
x-parameter: For a fax pass-through operation, the SDP fields of the re-INVITE requests and 200 OK responses contain X-fax description; for a modem pass-through operation, the SDP fields of the re-INVITE requests and 200 OK responses contain X-modem description. Examples # Configure the device to use the address in the To field as the address in the From field when sending a SIP request.
sip-comp server Use sip-comp server to configure the Server header field in SIP response messages. Use undo sip-comp server to remove the configuration. Syntax sip-comp server product-name product-version undo sip-comp server Default The Server header field in SIP response messages is not configured. Views SIP client view Default command level 2: System level Parameters server product-name product-version: Indicates the content of the Server header field in SIP response messages.
Parameters domain-name: Domain name of the SIP device. The value consists of 1 to 31 characters, which are not case-sensitive and include numbers 0 through 9, letters A through Z or a through z, underlines (_), hyphens (-), and dots (.). Examples # Set the domain name of the SIP device to hello.com. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] sip-domain hello.com source-bind Use source-bind to specify a source IP address for outbound SIP signaling or media flows.
Syntax timer connection age { tcp tcp-age-time | tls tls-age-time } undo timer connection age { tcp | tls } Default The aging time for TCP connections is 5 minutes, and that for TLS connections is 30 minutes. Views SIP client view Default command level 2: System level Parameters tcp tcp-age-time: Sets the aging time (in minutes) for TCP connections, in the range 5 to 30. If the idle time of an established TCP connection reaches the specified aging time, the connection will be closed.
Examples # Set the interval for the voice entity or SIP trunk account to re-register with the registrar after a registration failure to 300 seconds. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timer registration retry 300 timer registration expires Use timer registration expires to set the registration expiration time. Use undo timer registration expires to restore the default.
undo timer registration divider Default The registration percentage is 80%. Views SIP client view Default command level 2: System level Parameters percentage: Registration percentage, in the range 50% to 100%. Examples # Set the registration percentage to 50%.
Related commands • timer registration divider • timer registration expires timer session-expires Use timer session-expires to enable periodic refresh of SIP sessions and set the maximum and minimum session expiration time. Use undo timer session-expires to restore the default. Syntax timer session-expires seconds [ minimum min-seconds ] undo timer session-expires Default • The periodic refresh of SIP sessions is not enabled automatically.
Default The global transport layer protocol is UDP, and no transport layer protocol is specified for a VoIP voice entity. If the transport layer protocol is not specified for a VoIP voice entity, the global setting is applied. Views SIP client view, VoIP voice entity view Default command level 2: System level Parameters udp: Specifies UDP as the transport layer protocol for outgoing SIP calls. tcp: Specifies TCP as the transport layer protocol for outgoing SIP calls.
Views POTS voice entity view Default command level 2: System level Parameters user-info: Specifies a username. A username contains no more than 31 characters, and can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.). The total length of the username and the domain name cannot exceed 255 characters. domain domain-name: Specify the domain name. The domain name consists of character strings separated by a dot, for example, aabbcc.com.
Usage guidelines Executing the url command in SIP client view specifies the global SIP URL scheme. If you want to configure a different SIP URL scheme for an individual call, you can specify the SIP URL scheme in corresponding VoIP voice entity view. When the SIP URL scheme configured in VoIP voice entity view and that configured in SIP client view are different, the former is adopted. The VoIP voice entity configuration takes precedence over global configuration.
cnonce cnonce: Authentication information field used for handshake authentication between the registrar and the SIP UA, This field consists of a string of 1 to 50 case-sensitive characters. The characters double quotation mark (") and backward slash (\) are invalid. realm realm: Domain name used for handshake authentication between the registrar and SIP UA. The domain name consists of a string of 1 to 50 case-sensitive characters. The characters double quotation mark (") and backward slash (\) are invalid.
Default No VPN instance is configured for SIP. Views Voice view Default command level 2: System level Parameters vpn-instance-name: Specifies a VPN instance of MPLS L3VPN, a string of 1 to 31 case-sensitive characters. Usage guidelines You can configure this command only when no SIP services run. To use source IP address binding for SIP messages, the VPN instance associated with the source interface must be the same as the VPN instance specified in this command.
Examples # Enable fuzzy telephone number registration.
SIP local survival configuration commands area-prefix Use area-prefix to configure an area prefix. Use undo area-prefix to remove an area prefix or all area prefixes. Syntax area-prefix prefix undo area-prefix { prefix | all } Default No area prefix is configured. Views SIP server view Default command level 2: System level Parameters prefix: Area prefix, consisting of 1 to 15 digits. all: Removes all area prefixes.
Views Register user view Default command level 2: System level Parameters username username: Username used for authentication, consisting of 1 to 63 case-sensitive characters excluding backward slash (\) and double quotation marks ("). cipher: Specifies a ciphertext password. simple: Specifies a plaintext password. password: Specifies the password string. This argument is case-sensitive. If simple is specified, it must be a string of 1 to 16 characters.
call-rule-set Use call-rule-set to enter call rule set view. Syntax call-rule-set Views SIP server view Default command level 2: System level Examples # Enter call rule set view.
--------------------------------------------------------------------1 404 online 192.168.0.98:5060 2 325 offline 3 380 online 192.168.0.57:5060 Table 50 Command output Field Description user Tag of a user. number Directory number of a user. Registration status of a user: status • Offline. • Online. address IP address and port number that a user registers. display voice sip-server resource-statistic Use display voice sip-server resource-statistic to display server resource information.
SSA_Call 128 0 128 SSA_Sub 128 0 128 Table 51 Command output Field Description State of local SIP server: SIP Server state • Active. • Inactive. CbType Type of the resource control module. Total Total number of resource control modules. Used Number of used resource control modules. Free Number of free resource control modules. SLC_Conf Service logic control (SLC) control module. SLC_Call SLC call module. SLC_Sub SLC subscription module. SLC_Reg SLC registration module.
When the maximum registration interval configured on the voice gateway is greater than the maximum active time configured on the local SIP server, the maximum registration interval is subject to the one configured on the local SIP server. Examples # Set the maximum registration interval for user 1234 to 3,700 seconds.
number Use number to configure the directory number for a registered user. Use undo number to remove the configured directory number. Syntax number party-number undo number Default No directory number is configured for the user. Views Register user view Default command level 2: System level Parameters party-number: Directory number for a registered user, consisting of 1 to 31 digits. Examples # Configure the directory number 300 for registered user 1234.
port port-number: Port number of the remote server, in the range of 1 to 65535. The default port number is 5060. keepalive time-interval: Interval in seconds of sending OPTION messages to the remote server, in the range of 64 to 128. The default interval is 64 seconds. Usage guidelines If the local SIP server operates in the alive mode, you can configure the probe remote-server ipv4 command only when the local SIP server is disabled. Examples # Configure the keepalive probe.
Syntax rule tag { deny | permit } { incoming | outgoing } { pattern | any } undo rule { tag | all } Default No call rule is configured. Views Service view Default command level 2: System level Parameters tag: Call rule tag in the range of 0 to 31. deny: Denies calls. permit: Permits calls. incoming: Incoming calls. outgoing: Outgoing calls. pattern: Number pattern, consisting of digits and dots (.). Each dot represents a character and can only appear at the end of a number.
Parameters tag: Call rule set tag in the range of 0 to 31. Usage guidelines You can use the rule tag { permit | deny } { incoming | outgoing } pattern command in call rule view to set a call rule. Examples # Create a call rule.
[Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] server-bind ipv4 192.168.0.92 server enable Use server enable to enable the local SIP server. Use undo server enable to disable the local SIP server. Syntax server enable undo server enable Default The local SIP server is disabled. Views SIP server view Default command level 2: System level Usage guidelines The functions of the local SIP server can take effect only after you configure the server enable command.
Examples # Enter sip server view. system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] srs Use srs to apply call rule set. Use undo srs to remove the application. Syntax srs tag undo srs Default No call rule set is applied. Views SIP server view, register user view Default command level 2: System level Parameters tag: Call rule set tag in the range of 0 to 31. The call rule set corresponding to a tag must have been configured.
trunk Use trunk to configure a static route entry. Use undo trunk to delete a static route entry or all static route entries. Syntax trunk tag called-number called-pattern ipv4 dest-ip-addr [ port port-number ] [ area-prefix prefix ] undo trunk { tag | all } Default No call route entry is configured. Views Call route view Default command level 2: System level Parameters tag: Route entry tag in the range of 0 to 31. Each tag represents a route entry. At most 32 route entries can be configured.
Views SIP server view Default command level 2: System level Parameters ipv4 ipv4-address: IPv4 address of a trusted node. port port-number: Port number of a trusted node, in the range of 1 to 65535. The default port number is 5060. all: All trusted nodes. Usage guidelines At most eight trusted nodes can be specified on the local SIP server. Only an IP address, rather than a port number can specify a trusted node. Examples # Specify a trusted node by its IP address 100.1.1.125.
SIP trunk configuration commands address Use address to add a member server to a SIP server group and configure the server information. Use undo address to delete the configuration. Syntax address index-number { ipv4 ip-address | dns dns-name } [ port port-number ] [ transport { udp | tcp | tls } ] [ url { sip | sips } ] undo address index-number Default A SIP server group has no member server.
Usage guidelines An index represents the priority of a member server in the SIP server group. The smaller the index value, the higher the priority. You can add at most five member severs to a SIP server group. If an index already exists, the new configuration overwrites the existing one. Examples # Add member server 1 to SIP server group 1, and configure the server information: set the IPv4 address of the SIP server to 192.168.1.
address sip server-group Use address sip server-group to bind a SIP server group to a VoIP voice entity. Use undo address sip server-group to cancel the binding between a SIP server group and a VoIP voice entity. Syntax address sip server-group group-number undo address sip server-group Default A VoIP voice entity has no any SIP server group bound to it.
Default command level 2: System level Parameters contact-user user-name: Host username, a case-sensitive string of 1 to 31 characters excluding double quotation marks ("), backslash (\), or space. host-name host-name: Host name, a string of 1 to 255 characters, which are not case-sensitive. A host name can include letters, digits, hyphens (-), and underscores (_), and cannot include any space.
Examples # Disable SIP trunk account 2. system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2 [Sysname-voice-account-2] undo account enable bind sip-trunk account Use bind sip-trunk account to bind a SIP trunk account to a VoIP voice entity. Use undo bind sip-trunk account to cancel the binding between a SIP trunk account and a VoIP voice entity.
undo codec transcoding Default Codec transcoding is disabled. Views VoIP voice entity view Default command level 2: System level Usage guidelines The codec transcoding feature does not take effect in any of the following cases: • Codec transcoding is enabled, but no DSP resources are available for codec transcoding. • Codec transparent transfer is enabled. • Media flow-around is enabled. Examples # Enable codec transcoding on the SIP trunk device.
[Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] codec transparent description Use description to configure the description for a SIP server group. Use undo description to delete the description for a SIP server group. Syntax description text undo description Default A SIP server group has no description configured.
include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters. Examples # Display SIP trunk account information. display voice sip-trunk account ID User Group Server 1 1000 1 202.10.22.188:5060 2 2000 1 abc.com:5060 Exp 120 Status Online 400 Online Table 52 Command output Field Description ID SIP trunk account index. User Host username. Group SIP server group index.
include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters. Examples # Display the details of SIP server group 1. display voice server-group 1 The information of server group 1 Group name: ITSPA Description: ITSP A’s Proxy Server list Server list: Index 1: sip:192.169.0.1:5060;transport=udp Index 2: sips:abc.
Usage guidelines If codec transparent transfer or media flow-around is enabled, the early-offer forced command does not take effect. Examples # Enable DO-EO conversion on the SIP trunk device. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] early-offer forced Related commands • codec transparent • media flow-around group-name Use group-name to specify a name for a SIP server group.
Related commands • address • assign hot-swap enable Use hot-swap enable to enable the real-time switching function in a SIP server group. Use undo hot-swap enable to disable the real-time switching function in a SIP server group. Syntax hot-swap enable undo hot-swap enable Default The real-time switching function in a SIP server group is disabled. Views Server group view Default command level 2: System level Examples # Enable the real-time switching function in SIP server group 1.
interval seconds: Interval (in seconds) for sending OPTIONS messages to the SIP servers, in the range 5 to 65535. The default interval is 60 seconds. register: Specifies the REGISTER keep-alive mode. Usage guidelines With the keep-alive function enabled, the SIP trunk device selects the current server according to the detect result and the redundancy mode. If the keep-alive function is disabled, the current server is always the one with the highest priority in the SIP server group.
any of the above mentioned three header fields, the host name in the From header field of the INVITE message is used as the source host name. You can specify only one source host name prefix based match rule for a VoIP voice entity. If you execute match source host-prefix command multiple times, the new configuration overwrites the existing one. Examples # Specify that calls with a source host name starting with Bil are permitted on VoIP voice entity 1.
[Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 3 voip [Sysname-voice-dial-entity3] match destination host-prefix ali match source address Use match source address to match a source address for a VoIP voice entity. Use undo match source address to delete the call match rule that specifies the source address.
[Sysname-voice-dial-entity3] match source address ipv4 100.1.1.* media flow-around Use media flow-around to enable media flow-around on the SIP trunk device. This function enables the media packets to pass directly between two SIP endpoints, without the intervention of the SIP trunk device. Use undo media flow-around to restore the default. Syntax media flow-around undo media flow-around Default Media flow-around is disabled.
Default command level 2: System level Examples # Enable midcall signaling pass-through of SIP calls on the SIP trunk device. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip proxy server-group Use proxy server-group to specify a SIP server group to be used as the proxy server. Use undo proxy server-group to delete the proxy server configuration.
Views Account view Default command level 2: System level Parameters group-number: Index of the registrar bound to the SIP trunk account, in the range 1 to 10. expires seconds: Registration expiration interval of a SIP trunk account, in the range 60 to 3600, in seconds. If this parameter is not configured, the system applies the global registration expiration interval configured with the timer registration expires command in SIP client view. Usage guidelines The specified SIP server group must exist.
Usage guidelines To enable the registration function for a SIP trunk account, you need to assign it with a host username or associate it with a SIP server group. When the registration function for a SIP trunk account is enabled, you cannot change its host username or associated SIP server group. Examples # Assign 123 as the host name to SIP trunk account 2, and associate SIP trunk account 2 with SIP server group 2. Then, enable the registration function for the SIP trunk account.
Related commands keepalive server-group Use server-group to create a SIP server group and enter server group view. If the created server group already exits, use this command to enter server group view. Use undo server-group to delete one or all SIP server groups. Syntax server-group group-number undo server-group { group-number | all } Views Voice view Default command level 2: System level Parameters group-number: SIP server group index, in the range 1 to 10. all: Specifies all SIP server groups.
Default command level 2: System level Parameters account account-index: SIP trunk account index, in the range 1 to 16. all: Specifies all SIP trunk accounts. Usage guidelines A SIP trunk account that is bound to a SIP server group or a VoIP voice entity cannot be deleted. The undo sip-trunk account all command can be executed successfully only when all SIP trunk accounts are not bound to any SIP server group or a VoIP voice entity. Examples # Create SIP trunk account 2 and enter its view.
source-bind Use source-bind to specify a source IP address for sent SIP signaling or media flows. Use undo source-bind to remove the binding. Syntax source-bind { media | signal } { interface-type interface-number | ipv4 ip-address } undo source-bind { media | signal } Default No source address is bound for SIP signaling and media flows. Views Server group view Default command level 2: System level Parameters media: Media traffic. signal: Signaling traffic.
Usage guidelines If call forwarding is enabled, the SIP trunk device processes call forwarding information without notifying the calling party, and does not transfer call forwarding information to the endpoints.
Syntax user username password { cipher | simple } password undo user Default A SIP trunk account has no authentication username or password. Views Account view Default command level 2: System level Parameters username: SIP trunk account username used for registration authentication, a case-sensitive string of 1 to 63 characters. The characters double quotation mark (") and backward slash (\) are invalid. cipher: Specifies a ciphertext password. simple: Specifies a plaintext password.
H.323 configuration commands address Use address to configure the H.323 routing between a VoIP voice entity and a called party. Use undo address to remove the H.323 routing that has been configured. Syntax address { ip ip-address | ras } undo address { ip | ras } Default No routing policy is configured between the VoIP voice entity and the called party. Views VoIP voice entity view Default command level 2: System level Parameters ip ip-address: IP address of the called party.
Default No area ID is assigned to the H.323 gateway. Views Gatekeeper client view Default command level 2: System level Parameters string: Area ID, a string of 1 to 31 characters excluding spaces. The characters can be digits 0 through 9, or the pound sign (#) which can be used as a delimiter in the string.
Examples # Display the registration state information of the voice gateway.
Usage guidelines Use the quit command to exit gatekeeper client view. Examples # Enter gatekeeper client view. system-view [Sysname] voice-setup [Sysname-voice] gk-client [Sysname-voice-gk] Related commands • area-id • gk-2nd-id • gk-id • gw-address • gw-id • ras-on gk-2nd-id Use gk-2nd-id to configure the secondary gatekeeper for the gateway. Use undo gk-2nd-id to remove the secondary gatekeeper.
Examples # Configure a secondary gatekeeper, setting its IP address to 1.1.1.2, name to gk-backup, and RAS port to the default. system-view [Sysname] voice-setup [Sysname-voice] gk-client [Sysname-voice-gk] gk-id gk-center gk-addr 1.1.1.1 [Sysname-voice-gk] gk-2nd-id gk-backup gk-addr 1.1.1.2 Related commands gk-id gk-id Use gk-id to configure the primary gatekeeper for the gateway. Use undo gk-id to remove the primary gatekeeper.
system-view [Sysname] voice-setup [Sysname-voice] gk-client [Sysname-voice-gk] gk-id gk-center gk-addr 1.1.1.1 gk-security call enable Use gk-security call enable to enable security calling on the voice gateway. Use undo gk-security call enable to disable security calling on the voice gateway. Syntax gk-security call enable undo gk-security call enable Default Security calling is enabled.
Parameters cipher: Specifies a ciphertext password. simple: Specifies a plaintext password. password: Specifies the password string. This argument is case-sensitive. If simple is specified, it must be a string of 1 to 16 characters. If cipher is specified, it must be a string of 1 to 53 characters. Usage guidelines Messages exchanged during the entire registration process will carry the registration password, if configured on the voice gateway.
system-view [Sysname] voice-setup [Sysname-voice] gk-client [Sysname-voice-gk] gw-address 1.1.1.1 Related commands • area-id • gk-2nd-id • gk-id • gw-address • ras-on gw-id Use gw-id to assign an alias to the gateway. This alias overwrites the old one, if any. Use undo gw-id to remove the alias. Syntax gw-id namestring undo gw-id Default No alias is assigned to the gateway.
ras-on Use ras-on to activate gatekeeper client to register with the gatekeeper. Use undo ras-on to deactivate gatekeeper client. Syntax ras-on undo ras-on Default Gatekeeper client is disabled. Views Gatekeeper client view Default command level 2: System level Usage guidelines Only when the gatekeeper client function is active, the normal communication can be maintained between the voice gateway and the gatekeeper.
Parameters descriptor: H.323 descriptor, a case-sensitive string of 1 to 64 characters. Usage guidelines HP recommends that you use the default descriptor. If at both ends are the devices of HP, you are recommended to configure the same descriptor for them. Examples # Configure an H.323 descriptor mystring for the voice gateway.
Call services configuration commands backup-rule loose Use backup-rule loose to configure the call backup mode as loose. Use undo backup-rule loose to restore the default. Syntax backup-rule loose undo backup-rule loose Default The strict call backup mode is applied. Views Voice view Default command level 2: System level Examples # Configure the call backup mode as loose.
Parameters forward-number number: Specifies a forwarded-to number in the E.164 format, a string of 1 to 31 digits, 0 through 9. Examples # Enable call forwarding no reply for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
Related commands • call-forwarding no-reply enable • call-forwarding unavailable enable • call-forwarding unconditional enable • call-forwarding priority call-forwarding priority NOTE: This command applies only to the FXS voice subscriber line. Use call-forwarding priority to configure a priority level for call forwarding. Use undo call-forwarding priority to restore the default. Syntax call-forwarding priority level undo call-forwarding priority Default The call forwarding priority level is 2.
call-forwarding unavailable enable NOTE: This command applies only to the FXS voice subscriber line. Use call-forwarding unavailable enable to enable call forwarding unavailable. Use undo call-forwarding unavailable enable to restore the default. Syntax call-forwarding unavailable enable forward-number number undo call-forwarding unavailable enable Default Call forwarding unavailable is disabled.
Default Call forwarding unconditional is disabled. Views Voice subscriber line view Default command level 2: System level Parameters forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31 digits, 0 through 9. Examples # Enable call forwarding unconditional for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
[Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-hold enable call-hold-format Use call-hold-format to configure a tone playing mode for call hold. Use undo call-hold-format to restore the default. Syntax call-hold-format { inactive | sendonly [ media-play media-id | moh-number string ] } undo call-hold-format Default The tone playing mode is inactive, or silent mode. Views Voice view Default command level 2: System level Parameters inactive: Specifies the silent mode for call hold.
Syntax call-transfer enable undo call- transfer enable Default Call transfer is disabled. Views Voice subscriber line view Default command level 2: System level Usage guidelines Call hold must be enabled before call transfer. Examples # Enable call transfer for voice subscriber line 1/0.
Examples # Set the call transfer start delay to 2 seconds for voice subscriber line 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-transfer start-delay 2 Related commands call-transfer enable call-waiting Use call-waiting to configure parameters for a call waiting tone. Use undo call-waiting to restore the default.
Syntax call-waiting enable undo call-waiting enable Default Call waiting is disabled. Views Voice subscriber line view Default command level 2: System level Examples # Enable call waiting for voice subscriber line 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-waiting enable Related commands • call-waiting • call-waiting priority call-waiting priority Use call-waiting priority to configure a priority level for call waiting.
system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-waiting priority 1 Related commands • call-waiting • call-waiting priority conference enable Use conference enable to enable the three-party conference function for a voice subscriber line. Use undo conference enable to restore the default. Syntax conference enable undo conference enable Default The three-party conference function is disabled.
Views Voice subscriber line view Default command level 2: System level Examples # Enable incoming call barring for voice subscriber line 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] dialin-restriction enable dialout-restriction enable Use dialout-restriction enable to enable outgoing call barring for a voice subscriber line. Use undo dialout-restriction enable to restore the default.
display voice sip subscribe-state Use display voice sip subscribe-state to display the information of subscription, including phone numbers, subscription server address, effective time, and subscription state. Syntax display voice sip subscribe-state [ | { begin | exclude | include } regular-expression ] Views Any view Default command level 1: Monitor level Parameters |: Filters command output by specifying a regular expression.
Syntax display voice ss mwi { all | number number } [ | { begin | exclude | include } regular-expression ] Views Any view Default command level 1: Monitor level Parameters all: Displays the MWI information of all numbers. number number: Displays the MWI information of a specified number. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide.
Field Description Message waiting identifier: Messages-Waiting • Yes—There is/are waiting messages on the voice mailbox server. • No—There is no waiting message on the voice mailbox server. As shown in the above example, Messages-Waiting: Yes indicates that there are waiting messages in the mailbox of number 1515. Number of new messages/number of old messages (number of new urgent messages/ number of old urgent messages).
seconds: Specifies the door open duration in seconds. The value range is 30 to 300, and the default is 60. Usage guidelines The password is always saved to the configuration file in cipher text, whether you specify the cipher or simple keyword. If you execute this command multiple times, the most recent configuration takes effect. Install an SIC audio card on the device on which the door opening-enabled FXS voice subscriber line resides.
enable out-of-band NTE transmission, and you need to execute the outband nte command on the called entity to enable it. Examples # Enable the setting of the Feature service for voice subscriber line 1/0. system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] feature permit hunt-group enable Use hunt-group enable to enable hunt group. Use undo hunt-group enable to disable hunt group. Syntax hunt-group enable undo hunt-group enable Default Hunt group is disabled.
Views Voice subscriber line view Default command level 2: System level Parameters level: Hunt group priority level in the range of 1 to 3. The smaller the value, the higher the priority. Usage guidelines The priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively. The smaller the value, the higher the priority level. When you change the priority level of a feature, make sure different features have different priority levels.
Related commands distinguish-localtalk (in chapter "Voice entity configuration commands") mwi enable NOTE: This command is only applicable to the FXS voice subscriber line. Use mwi enable to enable MWI. Use undo mwi enable to disable MWI. Syntax mwi enable undo mwi enable Default MWI is disabled. Views Voice subscriber line view Default command level 2: System level Examples # Enable MWI for voice subscriber line 3/0.
Parameters length: Duration of playing the message waiting tone in seconds, in the range 1 to 60. Examples # Configure the duration of the message waiting tone as 4 seconds for voice subscriber line 3/0. system-view [Sysname] subscriber-line 3/0 [Sysname-subscriber-line3/0] mwi tone-duration 4 mwi-server Use mwi-server to configure the related information of the voice mailbox server. Use undo mwi-server to remove the configurations.
loose: Loose match, which indicates that strict consistency check is not needed, so the call ID that the NOTIFY is sent to can be different from the call ID that proposed the subscription. strict: Strict match, which indicates that strict consistency check is needed, so the call ID that the NOTIFY is sent to must be the same as the call ID that proposed the subscription. tcp: Specifies TCP as the transport layer protocol to be used during subscription.
undo timer called-hookon-delay Default Calling party control is disabled. The on-hook delay of the called party is set to 0. Views Analog FXS voice subscriber line view Default command level 2: System level Parameters seconds: Specifies the on-hook delay of the called party, in the range of 0 to 90 seconds. Examples # Enable calling party control on voice subscriber line 1/0 and set the on-hook delay time of the called party to 90 seconds.
Call-watch configuration commands The call-watch function is only applicable to voice E1/T1 interfaces. The E1/T1 interfaces mentioned in this document are all voice interfaces. call-watch group Use call-watch group to associate the current E1/T1 interface with a call-watch group in the specified mode. Use undo call-watch group to remove the association.
call-watch rule Use call-watch rule to create a call-watch monitoring rule in a call-watch group. If this rule is the first rule for the call-watch group, the group is created as a result. Use undo call-watch rule to delete the specified monitoring rule, or if no local interface or track object IP is specified, all monitoring rules, from a monitor group. The monitor group is deleted upon removal of the last rule.
display call-watch status Use display call-watch status to display information about the call-watch group associated with the specified E1/T1 interface. If no interface is specified, the information of all call-watch groups associated with an E1/T1 interface is displayed.
Field Description Call-Watch rule 2 local interface Ethernet1/1 Indicate that monitor group 2 monitors local interface Ethernet 1/1. network is DOWN The network layer state of the monitored local interfaces: up or down. It is down in this example.
Fax over IP configuration commands default entity fax Use default entity fax to set fax parameters to the default values globally. Use undo default entity fax to restore the fax parameters of the system to the defaults.
cng-switch enable: Enables CNG fax switchover. ecm: Enables the fax error correction mode. It is disabled by default. level level: Specifies the fax signal level in the range of –60 to –3 dBm. The default value is –15 dBm. local-train threshold threshold: Specifies the threshold percentage of fax local training, in the range of 0 to 100. The default value is 10. nsf-on: Enables NSF message transmission. It is disabled by default. protocol: Specifies the transport protocol of the fax. By default, the T.
Views Any view Default command level 2: System level Parameters |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression.
FAX_VOFR_MOTOROLA : 00:00:00 FAX_VOIP_STDT38 : 00:00:00 FAX_VOIP_T38 : 00:00:00 Processed Pages : 0 FAX_VOFR_STANDARD_SWITCH: 0 FAX_VOFR_FRF11_TRUNK : 0 FAX_VOFR_FRF11_SWITCH : 0 FAX_VOFR_MOTOROLA : 0 FAX_VOIP_STDT38 : 0 FAX_VOIP_T38 : 0 } Statistics about using fax baudrate: { V27 2400 : 0 V27 4800 : 0 V29 7200 : 0 V29 9600 : 0 V17 7200 : 0 V17 9600 : 0 V17 12000: 0 V17 14400: 0 } Statistics about using ECM or Non-ECM mode: { ECM : 0 Non-ECM: 0 } Statistics
Table 58 Command output Field Description FAX_VOFR_STANDARD_SWITCH Fax statistics for standard VoFR. FAX_VOFR_FRF11_TRUNK Fax statistics for FRF.11 trunk VoFR. FAX_VOFR_FRF11_SWITCH Fax statistics for FRF.11 switched VoFR. FAX_VOFR_MOTOROLA Fax statistics for Motorola compatible VoFR. FAX_VOIP_STDT38 Fax statistics for standard T.38 VoIP. FAX_VOIP_T38 Fax statistics for T.38 VoIP.
Syntax fax baudrate { 2400 | 4800 | 9600 | 14400 | disable | voice } undo fax baudrate Views POTS entity view, VoIP entity view, VoFR entity view Default command level 2: System level Parameters 2400: Sets the maximum fax baud rate to 2400 bps. 4800: Negotiates the fax baud rate first in accordance with the V.27 fax protocol. The maximum fax baud rate is 4800 bps. 9600: Negotiates the fax baud rate first in accordance with the V.29 fax protocol. The maximum fax baud rate is 9600 bps.
Default The CNG fax switchover function is disabled. Views POTS entity view, VoIP entity view Default command level 2: System level Examples # Enable the CNG fax switchover function. system-view [sysname] voice-setup [sysname-voice] dial-program [sysname-voice-dial] entity 100 pots [sysname-voice-dial-entity100] fax cng-switch enable fax ecm Use fax ecm to configure the ECM mode for the fax. Use undo fax ecm to restore the default.
fax level Use fax level to configure the transmit energy level of a gateway carrier. Use undo fax level to restore the default. Syntax fax level level undo fax level Default The transmit energy level of a gateway carrier is –15 dBm. Views POTS entity view, VoIP entity view, VoFR entity view Default command level 2: System level Parameters level: Level of the energy transmitted by a gateway carrier, namely the transmit energy level attenuation value in the range of –60 to –3 dBm.
Default command level 2: System level Parameters threshold: Local training threshold in percentage, in the range of 0 to 100. Usage guidelines Point-to-point training means the gateways do not participate in the rate training between two fax terminals. In this mode, rate training is performed between two fax terminals and is transparent to the gateways. For the point-to-point training, the gateway does not participate in rate training and the threshold is invalid.
fax protocol Use fax protocol to configure the type of protocol used for fax communication with other devices. Use undo fax protocol to restore the default type of protocol used for fax communication with other devices. Syntax fax protocol { t38 | standard-t38 } [ hb-redundancy number | lb-redundancy number ] fax protocol pcm { g711alaw | g711μlaw } undo fax protocol Default T.38 negotiation mode is used for fax. Views Voice entity view Default command level 2: System level Parameters t38: Uses T.
to a great extent and thereby, in the case of low bandwidth, affect the fax quality seriously. Therefore, the number of redundant packets should be selected correctly according to the network bandwidth. The pass-through mode is subject to such factors as loss of packet, jitter and delay, so the clocks on both communication sides must be kept synchronized. At present, only G.711 A-law and G.711 μ−law are supported, and the VAD function should be disabled.
modem compatible-param Use modem compatible-param to configure the value of NTE payload type for the NTE-compatible switching mode. Use undo modem compatible-param to restore the default. Syntax modem compatible-param payload-type undo modem compatible-param Default The value of the NTE payload type is 100.
Default command level 2: System level Parameters standard: Uses Re-Invite switching for Modem pass-through. nte-compatible: Uses NTE-compatible switching for Modem pass-through. g711alaw: Uses g711alaw codec for Modem pass-through. g711ulaw: Uses g711ulaw codec for Modem pass-through. Examples # Set the switching mode to NTE-compatible and the codec type to g711alaw for SIP Modem pass-through.
Views Voice view Default command level 2: System level Usage guidelines This command has global significance. The execution of this command can enable all voice entities to contain the T.38 faculty description in their faculty sets. Because NetMeeting does not support T.38 faculty description parsing, you must configure the voip h323-conf tcs-t38 command before interworking with NetMeeting.
IVR configuration commands call-normal Use call-normal to configure the normal secondary-call number match mode for the node. Use undo call-normal to remove the configuration. Syntax call-normal { length number-length | matching | terminator character } undo call-normal Default The match mode of normal secondary-call numbers is not configured. Views Call node view Default command level 2: System level Parameters length number-length: Matches the length of the numbers.
[Sysname-voice-ivr] node 1 call [Sysname-voice-ivr-node1] call-normal matching description Use description to configure the description string for the node. Use undo description to remove the configuration. Syntax description text undo description Default No node description string is configured. Views Call node view, Jump node view, Service node view Default command level 2: System level Parameters text: Node description string of 1 to 80 case-sensitive characters. Spaces are permitted.
Parameters string: Specifies the prefix code, a character string consisting of 1 to 31 characters that can include 0 through 9, pound sign (#), and asterisk (*). Table 59 describes these characters: Table 59 Description of characters in the string argument Character Meaning 0-9 Digits 0 through 9. Pound sign (#) or asterisk (*) Indicates a valid digit each. Usage guidelines When the number with a prefix exceeds 31 digits, only the first 31 digits are sent.
2 406 200 201 3 WAIT INPUT 3 606 300 301 6 CALL 4 806 400 401 9 IDLE Table 60 Command output Field Description Index Index of the call information. Called-Number Number of the called party. Caller-Number Number of the calling party. Entity IVR voice entity number of the called number. Node-Id Node ID. Current status: • IDLE—The node is idle. • PLAY MEDIA—The node is playing media files. • WAIT INPUT—The node is waiting for the input of the Status subscriber.
3 g711ulaw 1003 2 stop IP:100.1.1.1 4 g723r53 1004 2 stop IP:100.1.1.1 Table 61 Command output Field Description Index Playing index. Codec type, taking the following values: Codec • • • • g729r8. g711alaw. g711ulaw. g723r53. Media-Id Media resource file ID. Play-Times Play times of a file. Current status: Status • play. • stop. Current play type: Type • PSTN—The called party is from PSTN.
Codec Media-Id source Size (Bytes) Read-Number Cache-Number -------------------------------------------------------------------------g729r8 1000 cfa0:/wav/g7 69304 1 1 Real-Time 2 4 29r8/0.wav g711alaw 1006 MOH1/1 Table 62 Command output Field Description Codec Codec type of the media resource file. Media-Id Media resource file ID. Media source: • The file name is displayed if the media resource is a file.
Examples #Create IVR voice entity 100 and enter voice entity view. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 100 ivr extension Use extension to configure an extension secondary-call for a node. You can configure at most ten extension secondary-call numbers for a Call node. Use undo extension to remove the configuration.
undo input-error Default No input error processing method is configured for a node. Views Call node view, Jump node view Default command level 2: System level Parameters end-call: Terminates the call when the maximum number of input errors is reached. goto-pre-node: Return to the previous node when the maximum number of input errors is reached. goto-node node-id: Jumps to a specified node when the maximum number of input errors is reached.
# Configure the processing method for handling subscriber input errors for Jump node 1: • The node should jump to node 20 after the maximum number of times permitted for inputting errors is reached. • The media resource ID is 1002. • The node plays voice prompts three times. • The maximum number of input errors is 5.
[Sysname-voice-ivr] ivr-input-error media-play 10002 2 repeat 5 ivr-root Use ivr-root to specify the root node (the first node to be executed) of an IVR voice entity. Use undo ivr-root to remove the configuration. Syntax ivr-root node-id undo ivr-root Default The root node is not configured for an IVR voice entity. Views IVR voice entity view Default command level 2: System level Parameters node-id: Specifies the ID of the root node, in the range 1 to 256.
[Sysname-voice-ivr] ivr-timeout Use ivr-timeout to configure the IVR global input-timeout processing method. Use undo ivr-timeout to restore the default. Syntax ivr-timeout { expires seconds | media-play media-id [ play-times ] | repeat repeat-times } * undo ivr-timeout Default The timeout time is 10 seconds, and the maximum timeout times are 3. The system does not play voice prompts for input timeouts and terminates the call after the maximum number of times is reached.
Views IVR management view Default command level 2: System level Parameters g711alaw: Enters g711alaw codec view. g711ulaw: Enters g711ulaw codec view. g723r53: Enters g723r53 codec view. g729r8: Enters g729r8 codec view. Related commands • ivr-system • set-media Examples # Enter g729r8 codec view.
play-times: Specifies the times for playing voice prompts. The value range is 1 to 255, and the default is 1. force: Specifies that the subscriber can press the key only after the play of voice prompts is finished, otherwise, subscriber's input is considered invalid. Examples # Specify the node to play the audio file 10000 three times to the subscriber when waiting for the subscriber to press keys.
Use undo operation to remove the configuration. Syntax operation number { call-immediate call-number | end-call | goto-node node-id | goto-pre-node | media-play media-id [ play-times ] } undo operation number Default No function is configured for a Service node. Views Service node view Default command level 2: System level Parameters number: Specifies the serial number of the configured function, in the range 1 to 3. call-immediate call-number: Indicates immediate secondary-call.
Syntax select-rule operation-order 1st-operation 2nd-operation 3rd-operation undo select-rule operation-order Default The execution order is select-rule operation-order 1 2 3. Views Service node view Default command level 2: System level Parameters 1st-operation: Specifies the serial number of the function to be executed first, in the range of 1 to 3. 2nd-operation: Specifies the serial number of the function to be executed secondly.
Parameters media-id: Specifies the media resource file ID, in the range 1000 to 2147483647. file filename: Media resource file name. Spaces are permitted, and the file name must be in double-quote marks. The maximum length of the value is 136 bytes, excluding the length of double-quote marks. moh-interface interface-number: MOH audio input port number. all: All media resource file IDs. Examples # Specify 10001 as the media resource ID of the media resource file cfa0:/g729/ring.wav.
play-times: Specifies the times for playing voice prompts. The value range is 1 to 255, and the default is 1. repeat repeat-times: Specifies the maximum number of input timeouts. After an input timeout occurs, the node is executed again. When the maximum number of input timeouts is reached, the system terminates the call. The value range for repeat-times is 0 to 255, and the default is 3.
[Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] user-input 0 end-call 323
VoFR configuration commands address Use address to configure a channel to the peer voice gateway. Use undo address to remove the configuration. Syntax address { vofr-dynamic serial interface-number dlci-number | vofr-static serial interface-number dlci-number cid-number } undo address { vofr-dynamic | vofr-static } Default No channel to the peer voice gateway is configured.
• vofr • trunk-id • display fr vofr-info call-mode Use call-mode to configure the mode in which calls between the VoFR entity and the peer voice entity are established. Use undo call-mode to restore the default call mode. Syntax call-mode { dynamic | static } undo call-mode Default The dynamic mode is adopted. Views VoFR entity view Default command level 2: System level Parameters dynamic: Adopts the dynamic call mode. static: Adopts the FRF.11 trunk mode.
cid select-mode Use cid select-mode to configure the CID selection mode which the originating side of a VoFR call adopts. Use undo cid select-mode to restore the default. Syntax cid select-mode { max-poll | min-poll } undo cid select-mode Default CIDs are cyclically selected in descending order. Views Interface DLCI view Default command level 2: System level Parameters max-poll: Selects circuit IDs cyclically in descending order. min-poll: Selects circuit IDs cyclically in ascending order.
Views Any view Default command level 2: Monitor level Parameters serial interface-number: Displays the FRF.11 sub-channel information on a specified interface. dlci-number: Virtual circuit number in the range of 16 to 1007 |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow.
undo entity { entity-number | all | vofr } Views Voice dial program view Default command level 2: System level Parameters entity-number: Entity number in the range of 1 to 2147483647. all: All types of voice entities, including VoIP, POTS, VoFR, and IVR voice entities. vofr: VoFR voice entity. Usage guidelines When you configure VoIP entities, POTS entities, VoFR entities, and IVR entities, they must be identified with different entity-number.
seq-number Use seq-number to configure the VoFR packets sent by the local voice gateway to carry a sequence number. Use undo seq-number to restore the default. Syntax seq-number undo seq-number Default The VoFR packets sent by the local voice gateway do not carry any sequence number. Views VoFR entity view Default command level 2: System level Usage guidelines Usually, the configuration of the originating voice gateway determines whether VoFR packets carry a sequence number.
Default The VoFR packets sent by the local voice gateway do not carry any timestamp. Views VoFR entity view Default command level 2: System level Examples # Configure voice packets sent by VoFR entity 10 to carry a timestamp. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 vofr [Sysname-voice-dial-entity10] timestamp trunk-id Use trunk-id to configure a PSTN-dialed number in the FRF.11 trunk mode. Use undo trunk-id to restore the default.
voice bandwidth Use voice bandwidth to reserve a VoFR voice bandwidth. Use undo voice bandwidth to remove the reserved bandwidth. Syntax voice bandwidth reserved-bps [ reserved ] undo voice bandwidth Default No bandwidth is reserved for voice. Views Frame relay class view Default command level 2: System level Parameters reserved-bps: Reserved voice bandwidth in the range of 8000 to 45000000 bps. reserved: Reserves a VoFR voice bandwidth.
Parameters huawei-compatible: Adopts the Huawei-compatible mode. motorola-compatible: Adopts the Motorola-compatible mode for compatibility with VoFR of Motorola routers. The FRF.11 trunk mode does not support the Motorola-compatible protocol. dce: Specifies the virtual circuit to serve as a DCE in compliance with Annex G. dte: Specifies the virtual circuit to serve as a DTE in compliance with Annex G.
Use undo vofr frf11-timer to restore the default. Syntax vofr frf11-timer time undo vofr frf11-timer Default The trunk wait timer length is 30 seconds. Views Voice view Default command level 2: System level Parameters time: Trunk Wait timer length in the FRF.11 trunk mode, in the range of 10 to 600 seconds. Usage guidelines This command has global significance. The configuration is valid for all FRF.11 trunk calls after the command is executed. The Trunk Wait timer is specific to the FRF.
Voice RADIUS configuration commands aaa-client Use aaa-client to enter voice AAA client view. Syntax aaa-client Views Voice view Default command level 2: System level Examples # Enter voice AAA client view. system-view [Sysname] voice-setup [Sysname-voice] aaa-client [Sysname-voice-aaa] accounting Use accounting to enable the RADIUS accounting function for users who dial some access number. Use undo accounting to disable the RADIUS accounting function.
Examples # Enable the RADIUS accounting function for users who dial the access number 17909. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] accounting # Disable the RADIUS accounting function for users who dial the access number 17909.
Related commands • acct-method • accounting acct-method Use acct-method to configure an accounting method for the RADIUS client. Use undo acct-method to restore the default. Syntax acct-method { start-ack | start-no-ack | stop-only } undo acct-method Default The accounting method is start-no-ack. Views Voice AAA client view Default command level 2: System level Parameters start-ack: When the call setup begins, the voice gateway sends an Accounting-Start request to the RADIUS server.
authentication Use authentication to enable the RADIUS authentication function for users who dial some access number. Use undo authentication to disable the RADIUS authentication function. Syntax authentication undo authentication Default The RADIUS authentication function is disabled for users who dial access numbers. Views Access number view Default command level 2: System level Usage guidelines For each access number, you can specify the RADIUS server to perform authentication for users who dial it.
Default The authentication function is disabled for all one-stage dialing users. Views Voice AAA client view Default command level 2: System level Usage guidelines This command is applicable to only one-stage dialing users, instead of two-stage dialing users. With this function enabled, the calling number of one-stage dialing users who want to make IP calls is sent to the RADIUS server for authentication. Only users who pass authentication can make IP calls.
Enable the authentication function (by using the authentication command) before the authorization function. Otherwise, the authorization command is unavailable. Examples # Enable the authorization function for users who dial the access number 17909.
# Disable the authorization function for one-stage dialing users. [Sysname-voice-aaa] undo authorization-did Related commands authentication-did callednumber receive-method Use callednumber receive-method to configure the method of collecting digits of a called number. Use undo callednumber receive-method to restore the default.
card-digit Use card-digit to configure the number of digits in a card number for some access number in the card number/password process. Use undo card-digit to restore the default. Syntax card-digit card-digit undo card-digit Default The number of digits in a card number is 12 only when an access number is already configured for the card number/password process (by using the process-config command).
Syntax cdr { buffer size-number | duration time-length | threshold percentage } undo cdr { all | buffer | duration | threshold } Views Voice AAA client view Default command level 2: System level Parameters buffer size-number: Specifies the number of CDRs that can be saved in the buffer. The value range for the size-number argument is 0 to 500, and the default is 50. The value "0" indicates that no CDR can be saved. duration time-length: Specifies the lifetime of CDRs in seconds.
display voice access-number Use display voice access-number to display the configuration information and access numbers in voice AAA client view. Syntax display voice access-number [ | { begin | exclude | include } regular-expression ] Views Any view Default command level 2: System level Parameters |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide.
callednum receive = card digit termintor = password digit = 6 redialing times = 2 12 access number: [ 201 ] dialing process = voice-caller accounting = authentication = authorization callednum receive redialing times = language selected off off = off = immediate 2 = Chinese Table 64 Command output Field accounting-method Description Accounting method: start-ack, start-no-ack, or stop-only. See the acct-method command. Accounting function for one-stage dialing users.
Field Description Accounting function for two-stage dialing users. accounting • on—Enabled. • off—Disabled. See the accounting command. Authentication function for two-stage dialing users. authentication • on—Enabled. • off—Disabled. See the authentication command. Authorization function for two-stage dialing users. authorization • on—Enabled. • off—Disabled. See the authorization command. callednum receive card digit Method of collecting digits of a called number: terminator or immediate.
Parameters all: Displays all call records. callednumber called-number: Displays call records by called number. The called-number argument is a string of up to 31 characters, consisting of digits 0 through 9 and the asterisk *. callingnumber calling-number: Displays call records by calling number. The calling-number argument is a string of up to 31 characters, consisting of digits 0 through 9 and the asterisk *. card card-number: Displays call records by prepaid card number.
ReleaseTime = Oct 20, 2006 16:45:47 SendPackets = 71 packages SendBytes = 2982 bytes ReceivePackets = 111 packages ReceiveBytes = 4662 bytes Outgoing call leg [ 0 ]: CalledNumber = 2000 CallDuration = 00h 00m 02s EncodeType = G729R8 DecodeType = G729R8 ReleaseCause = Called hook on SignalType = SIP IpAddress/Port = 1.1.1.
Field Description Outgoing call leg [ 0 ] Information of the outgoing call leg. One call might involves multiple outgoing call legs. [ 0 ] identifies one outgoing call leg. CalledNumber Called number. CallDuration Call duration. EncodeType Encoding type. DecodeType Decoding type. ReleaseCause Call release cause. SignalType Signaling protocol (for example, R2, E&M, and H.323). VoiceInterface Voice interface. IpAddress/Port IP address and port number. SetupTime Call setup time.
Examples # Display statistics of messages exchanged between the voice RADIUS module, CMC module, and AAA module.
Field Description Leaving Leaving message. AAA=>VORDS: Messages from the AAA module to the voice RADIUS module. Authen_Accept Authentication_Accept message. Authen_Reject Authentication_Reject message. Author_Accept Authorization_Accept message. Author_Reject Authorization_Reject message. AcctRsp_PstnCaller Accounting_Response message for PSTN caller. AcctRsp_VoipCaller Accounting_Response message for VoIP caller. AcctRsp_PstnCalled Accounting_Response message for PSTN callee.
Default command level 2: System level Parameters access-number: Access number (for example, 169 and 17909), a string of up to 31 characters consisting of digits 0 through 9 and the wildcard dot (.). The wildcard dot (.) represents a digital character and must follow a digit or appear separately. all: Deletes all access numbers. Usage guidelines When you delete all configured access numbers, the voice gateway gives alarm information, requiring you to make a confirmation.
password-digit Use password-digit to configure the number of digits in a password for some access number in the card number/password process. Use undo password-digit to restore the default number of digits in a password for some access number in the card number/password process. Syntax password-digit password-digit undo password-digit Default The number of digits in a password for some access number in the card number/password process is 6.
Syntax process-config { callernumber | cardnumber | voice-caller } undo process-config Default The caller number process with IVR is specified for all access numbers. Views Access number view Default command level 2: System level Parameters callernumber: Specifies the two stage-dialing process as caller number process. After a user dials an access number, the voice gateway continues to play dial tones, prompting for a called number.
Examples # Specify the dialing process for the access number 17909 as card number/password process. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] process-config cardnumber # Restore the default dialing process for the access number 17909.
For the caller number process with IVR, you can use the redialtimes command to set times of reselecting a language and times of redialing a called number. Examples # Set the number of redial attempts to 4 for the access number 17909.
Views Access number view Default command level 2: System level Parameters enable: Enables the language selection function so that users can select a language to play prompt tones. chinese: Plays prompt tones in Chinese. english: Plays prompt tones in English. Usage guidelines This command is available only in the caller number process with IVR. Examples # Configure the voice gateway to play prompt tones in English.
Usage guidelines A timer resets every time the user dials a digit until all the digits are dialed. If the timer times out before the dialing finishes, there are two scenarios: • In the card number/password process and caller number process with IVR, if the number of redial attempts is not reached, the user is prompted to redial the number • In the caller number process, or if the number of redial attempts is reached, the user is prompted to hang up, and the call ends.
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 ABCDEFGHIJKLMNOPQRSTUVW call-fallback,177 A call-forwarding no-reply enable,267 aaa-client,334 call-forwarding on-busy enable,268 account enable,237 call-forwarding priority,269 accounting,334 call-forwarding unavailable enable,270 accounting-did,335 call-forwarding unconditional enable,270 acct-method,336 call-history,1 address,257 call-hold enable,271 address,234 call-hold-format,272 address,324 calling-name,58 address sip,176 callmode,111 address sip server-group,236 call-mode
cptone country-type,63 display voice ivr media-source,310 cptone tone-type,65 display voice jitter-buffer,23 crypto,177 display voice number-substitute,154 D display voice radius statistic,348 display voice server-group,241 default,66 display voice sip call-statistics,178 default entity compression,7 display voice sip connection,181 default entity fax,292 display voice sip dns-record,182 default entity payload-size,9 display voice sip dynamic-contact-address,183 default entity vad-on,10 dis
ivr-input-error,314 enable snmp trap updown,119 entity,33 ivr-root,315 entity ivr,311 ivr-system,315 entity vofr,327 ivr-timeout,316 enum-group,190 J expires,224 jitter-buffer delay,36 extension,312 jitter-buffer mode,35 F joined-conference enable,283 fast-connect,34 K fax baudrate,296 keepalive,244 fax cng-switch enable,297 keepalive,36 fax ecm,298 keepalive,190 fax level,299 fax local-train threshold,299 L fax nsf-on,300 line,37 fax protocol,301 line,122 fax train-mode,302 li
node,318 remote-party-id,203 number,226 renew,132 number-match,159 reset voice cmc statistic,93 number-priority,160 reset voice cmc statistic,42 number-substitute,161 reset voice fax statistics,304 O reset voice ipp statistic,43 reset voice ipp statistic,93 open-trunk,91 reset voice iva statistic,43 operation,318 reset voice iva statistic,93 outband,40 reset voice radius statistic,355 outband sip,193 reset voice sip connection,203 outband vofr,328 reset voice sip dns-record,204 outboun
sip,205 timer ring-back,101 sip-comp,206 timer session-expires,213 sip-comp agent,207 timer two-stage dial-interval,356 sip-comp server,208 timer wait-digit,102 sip-domain,208 timeslot-set,144 sip-server,230 timestamp,329 sip-trunk account,252 transmit gain,102 sip-trunk enable,253 transport,213 slic-gain,97 trunk,232 source-bind,209 trunk-direction,145 source-bind,254 trunk-id,330 special-character,137 trusted-point,232 srs,231 ts,146 subscriber-group,171 tunnel-on,46 subscriber
